Re: [Asterisk-Users] How to get Call Details Records

[CW_ASN]

Title: Message



Please don't cross message between 
lists.


  - Original Message - 
  From: 
  Mayank Mishra 
  To: [EMAIL PROTECTED] 
  
  Sent: Saturday, September 25, 2004 6:40 
  AM
  Subject: [Asterisk-Users] How to get Call 
  Details Records
  
  HI,
  Can anyone please 
  tell me
  
  1) Where does 
  asterisk store the call detail records?
  2) What is 
  thestructure of these call details records?
  2)How to 
  access the call detail records by any external 
application?
  
  Thanks in 
  advance
  Regards,
  Mayank
  
  

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Re: [Asterisk-Users] Whoa.... I'm owned but found ??

[CW_ASN]

Don't cross messages between lists.

Anyway, be more specific.


  - Original Message - 
  From: 
  shabanip 
  To: [EMAIL PROTECTED] 
  
  Sent: Saturday, September 25, 2004 12:02 
  PM
  Subject: [Asterisk-Users] Whoa I'm 
  owned but found ??
  
  I get this message at CLI.
  what does it mean?
  
  - shabanip
  
  

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Re: [Asterisk-Users] voicemail setup guide?

[CW_ASN]

 
 is there a well-written, easy to follow, voicemail setup guide for
 asterisk?

No, but you don't need setup guide. See wiki.

Regards,

Gus



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Re: [Asterisk-Users] Re: Numbering Plan and Siemens EWSD

[CW_ASN]

 Trace from their analyzer attached.

Can they send an EWSD trace???


 switchtype was already set to euroisdn, so that shouldn't be the problem.

 I first configured pridialplan=unknown, but the telecom partner asked me
 to change the TON (type of number) to unknown, and the NPI to
 ISDN/Telephony Numbering Plan (E.164/E.163).
A smart technician must avoid to use TON=Unknown.
Correct, E164 must be used.


 Setting the pridialplan to local was not allowed (a TON of
 'subscriber number' wouldn't work on their switch).
Bad data in tables, I presume... or you are sending crap.


 So I changed some code and added a PRI_PROVIDER constant 0x01 to libpri.h
 (TON: unknown, NPI: ISDN/Telephony NP), like they wanted.


 Bruno, so you're just using pridialplan=local / national ?
 Do you also use the prilocaldialplan ?


 The guys from Siemens told me that it was highly uncommon to connect a
 softswitch directly to the Siemens EWSD.
Softwitch? What softswitch? For EWSD, asterisk it's just a PBX, because is
connected thru PRI!


 Our telecom partner in Belgium is TTG / Ventelo, and we are the only
 ones who connect a softswitch to their Siemens.


 If anyone has some info, please let me know.


 Thanks in advance.



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Re: [Asterisk-Users] Numbering Plan and Siemens EWSD

[CW_ASN]

 Hi all,

 We're trying to hook up our Asterisk config (Card: TE410P) with a
 Siemens EWSD switch. The link is ok on both ends (green), with no errors.

 The problem is when we try to make a call from our side (via call
 files), we get the pri/E1 error
Ext: 1  Cause: Temporary failure (41), class = Network Congestion (2)

 Our Telecom partner (they checked with Siemens) mentioned that we need
 to configure a dialplan as

  numbering plan (Rec. E.164)
 The  stands for ISDN (Telephony), ISDN (Speech), etc

 This is what they told us, but the closest we can configure in Asterisk
 is the pridialplan (unknown, private, local, national, international).

 We tried all of them, with no difference.
 We also tried them with callerid set, no advance.


 Anyone familiar with this other dialplan, or with the integration of
 Asterisk/E1 with a Siemens EWSD switch.


 pri debug log of the call below (this was with pridialplan set to
'unknown')
 and without callerid.

 -- Making new call for cr 32780
  Protocol Discriminator: Q.931 (8)  len=32
  Call Ref: len= 2 (reference 12/0xC) (Originator)
  Message type: SETUP (5)
  Bearer Capability (len= 3) [ Ext: 1  Q.931 Std: 0  Info transfer
capability: Speech (0)
   Ext: 1  Trans mode/rate: 64kbps,
circuit-mode (16)
   Ext: 1  User information layer 1: A-Law
(35)
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
Dchan:
 0


As Bruno said, check that you are using euroisdn.

If you are not using ISDN equipment to dial thru pri, SETUP message is
wrong.

And please... change you pridial to local.
  Called Number (len=12) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
Unknown Number Plan (0) 'x' ]

Anyway, can you send a SIGNTRAC, or maybe a LTGTRAC (better to view more
deeply)?

Regards,

Gus



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Re: [Asterisk-Users] isdn cli

[CW_ASN]

 hi!

 I need to pass the CLI for my outgoing ISDN PRI call from * box.

 here's the ISDN protocol debug.

 Q.931
 Calling Number (len=10) [ Ext: 0  TON: Unknown Number Type (0)  NPI:
Unknown Number Plan (0)
   Presentation: Presentation permitted, user
 number passed network screening (1) '123123' ]
  Called Number (len=12) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
 Unknown Number Plan (0) '818818' ]
 Sending Complete (len= 0)

 But the CLI is not seen at the end mobile. Instead a fix number is seen.
Is this a problem with * ISDN  driivers ?

 but123

No, I cannot see a problem here...
Welcome to the PSTN world. Please read the past messages of this list. You
cannot send some number as you wish. It must be according with your DID, and
in the range specified for your provider.
Also, you provider must give you ANI for internal number or AMA for
internal number.

Regards,

Gus

P.D: I'm tired of this: TON: Unknown. It must be local.




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Re: [Asterisk-Users] Re: Re: MYSQL_FRIENDS and IAX problem

[CW_ASN]

 I believe that 'ast_data' is the solution to this problem, and will
 probably obsolete mysql friends.  However, I could be incorrect in that
 manner.  There are folks on this list who would be much better informed to
 say whether or not it will obsolete mysql friends.

 -Chris


I did not tests with iaxfriends, but I tested some with sipfriends. I'm
afraid that the support for sipfriends is not complete, because AFAIK, the
additional parameters of friend can't be set, such as defaultip, nat,
pickupgroup or callgroup. I dont know if ast_data bring some solution to
this.

Regards,

Gus



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Re: [Asterisk-Users] Adding voice mail box

[CW_ASN]

 Hi,

 I've forgotten the command to add a vm box, and searching google and wiki
I'm
 surpriced I cannot find it. I'd love to know where this is written, so I
can
 see how I managed to miss it!

 - -- 
 Steve

Look for your controb/script directory. The script is called 'addmailbox'.

Regards,

Gus



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Re: [Asterisk-Users] How to uninstall Asterisk?

[CW_ASN]

Rui:

IMHO=In My Humble Opinion

Regards,

Gus

- Original Message - 
From: ruixun wu [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 14, 2004 11:07 AM
Subject: Re: [Asterisk-Users] How to uninstall Asterisk?


 hi Gus and Roger,
Thanks for you reply. I choose no load the
 chan_oh323. The asterisk now can start again. :)
And Gus, could you tell me what's the meaning of
 IMHO? I can't find the topic about IMHO in WIFI.
 
 Thanks a lot!
 Best Regards
 Rui
 
 
 
 --- CW_ASN [EMAIL PROTECTED] wrote:   Hi,
  After I install openh323, the asterisk cann't
  work
   anymore. Asterisk failed in loading chan_oh323. I
   cann't deleted the openh323 package, so the only
  thing
   I can do is to reinstall Asterisk. I checked out
  the
   asterisk and make install Astersik without
  installed
   openh323, but when I started Asterisk, Asterisk
  still
   loaded openh323 and failed.
  Does anyone know how to uninstall Asterisk?
  
  If you don't like to load a channel or module, you
  can choose for two
  methods:
  - You can delete it. The channels and apps are
  located in
  /usr/lib/asterisk/modules.
  - You can choose to not load when asterisk loads.
  Use modules.conf, set
  noload = foo.so
  
  At least is strage...  I'm using chan_oh323 without
  failures, and IMHO, it's
  more stable and powerful than others. I'm not wish
  to start a war, it's just
  my opinion.
  
  Regards,
  
  Gus
  
  
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Re: [Asterisk-Users] How to uninstall Asterisk?

[CW_ASN]

Or HUMBLE? Maybe some native english guy can tell us...

E' lo mismo... ;)

- Original Message - 
From: Sebastian Nocetti [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, July 14, 2004 11:45 AM
Subject: RE: [Asterisk-Users] How to uninstall Asterisk?


 IN MY HONEST OPINION... IMHO

 I am right?


 -Mensaje original-
 De: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] En nombre de ruixun wu
 Enviado el: Miércoles, 14 de Julio de 2004 11:07 a.m.
 Para: [EMAIL PROTECTED]
 Asunto: Re: [Asterisk-Users] How to uninstall Asterisk?

 hi Gus and Roger,
Thanks for you reply. I choose no load the chan_oh323. The asterisk now
 can start again. :)
And Gus, could you tell me what's the meaning of IMHO? I can't find the
 topic about IMHO in WIFI.

 Thanks a lot!
 Best Regards
 Rui



 --- CW_ASN [EMAIL PROTECTED] wrote:   Hi,
  After I install openh323, the asterisk cann't
  work
   anymore. Asterisk failed in loading chan_oh323. I cann't deleted the
   openh323 package, so the only
  thing
   I can do is to reinstall Asterisk. I checked out
  the
   asterisk and make install Astersik without
  installed
   openh323, but when I started Asterisk, Asterisk
  still
   loaded openh323 and failed.
  Does anyone know how to uninstall Asterisk?
 
  If you don't like to load a channel or module, you can choose for two
  methods:
  - You can delete it. The channels and apps are located in
  /usr/lib/asterisk/modules.
  - You can choose to not load when asterisk loads.
  Use modules.conf, set
  noload = foo.so
 
  At least is strage...  I'm using chan_oh323 without failures, and
  IMHO, it's more stable and powerful than others. I'm not wish to start
  a war, it's just my opinion.
 
  Regards,
 
  Gus
 
 
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Re: [Asterisk-Users] New Asterisk bounty: SIP simultaneous

[CW_ASN]

 As I explained to you before we use it for our customer service in call
(B center and implemented in many call centres which really makes $.
(B
(BAll this stuff to do a simple call queue system??? Man, You need to read
(Bwiki. Read agents.conf and queue.conf before to begin a war here...
(BAll you need to do can be achieved with app_queue.
(B
(BRegards,
(B
(BGus
(B
(B
(B
(B___
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Re: [Asterisk-Users] Rotary phones? (No, I'm serious)

[CW_ASN]

Check wiki for patch... maybe it's you best option.

Regards,

Gus

- Original Message - 
From: Ethan [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, July 13, 2004 4:22 PM
Subject: [Asterisk-Users] Rotary phones? (No, I'm serious)


 
 Will the FXS cards that work with asterisk handle rotary? Are there any
 channel banks that can convert rotary to touch tone (like some sorta
 bridge)?
 
 The goal is to be able to log input from rotary phones. Full PBX
 functionality would be nice but...
 
 (It's for a project, not for serious production).
 
 --
// Ethan O'Toole
// http://users.757.org/~ethan
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Re: [Asterisk-Users] How to uninstall Asterisk?

[CW_ASN]

 Hi,
After I install openh323, the asterisk cann't work
 anymore. Asterisk failed in loading chan_oh323. I
 cann't deleted the openh323 package, so the only thing
 I can do is to reinstall Asterisk. I checked out the
 asterisk and make install Astersik without installed
 openh323, but when I started Asterisk, Asterisk still
 loaded openh323 and failed.
Does anyone know how to uninstall Asterisk?

If you don't like to load a channel or module, you can choose for two
methods:
- You can delete it. The channels and apps are located in
/usr/lib/asterisk/modules.
- You can choose to not load when asterisk loads. Use modules.conf, set
noload = foo.so

At least is strage...  I'm using chan_oh323 without failures, and IMHO, it's
more stable and powerful than others. I'm not wish to start a war, it's just
my opinion.

Regards,

Gus


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Re: [Asterisk-Users] Asterisk Queue Question

[CW_ASN]

 Is there any way for me to add myself to a call queue from outside of my
 Asterisk Box?

 For example,

 I have a queue set up on my asterisk box, and I want to call it on my Cell
 Phone, then add myself to the queue and hang up.. When a call comes into
the
 queue, I want it to be forwarded to my cell phone.

 Is this possible?
 I haven't been able to find info on it anywhere, but maybe I'm not looking
 in the right help..

Yes, exactly... use google o wiki.

The solution is AgentCallbackLogin.

  -= Info about application 'AgentCallbackLogin' =-

[Synopsis]:
Call agent callback login

[Description]:
  AgentCallbackLogin([AgentNo][|[EMAIL PROTECTED]):
Asks the agent to login to the system with callback.  Always returns -1.
The agent's callback extension is called (optionally with the specified
context.


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Re: Re[2]: [Asterisk-Users] Patch for call queues?

[CW_ASN]

It's included in CVS. I'm using it from there!
Anyway, the patch is 214. Look
http://bugs.digium.com/bug_view_page.php?bug_id=214

Regards,

Gus


 At 00:35:41, CW_ASN wrote:
  Please try CVS, AFAIK patch 214 doesn't included in stable branch.

 But I need to apply some other patches too that isn't included in the
 CVS! How can I do that when I install * CVS?

 Best regards,

 Robin



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Re: [Asterisk-Users] Anyone using gr303?

[CW_ASN]

 Anyone have any experience using gr303?

 May have a need to interface * to a Siemens Class-5 CO for pstn
 trunking (inbound and outbound).

 Rich


I assume Siemens Class5=EWSD.
EWSD is compatible with GR.303, and AFAIK it works with special national
project.
Which software version (APS) and which GP (LTG) modules (RAM qty) has your
EWSD? I can check it with Project Handbook.


Regards,

Gus



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Re: [Asterisk-Users] prepaid application

[CW_ASN]

Man, just provide us more info... debugs, logs, anything.
You don't need to pay for help.

Regards,

Gus


- Original Message - 
From: Stuart Baggs [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 30, 2004 6:57 PM
Subject: Re: [Asterisk-Users] prepaid application


 Could anyone install this on my * server for me? iw ill pay you $20

 - Original Message -
 From: Doug Harris [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Wednesday, June 30, 2004 10:30 PM
 Subject: RE: [Asterisk-Users] prepaid application


  how could any prepaid application be good if it does not update the
 balance
  :)
 
   -Original Message-
   From: [EMAIL PROTECTED]
   [mailto:[EMAIL PROTECTED] Behalf Of Hekuran
Doli
   Sent: Wednesday, June 30, 2004 9:58 AM
   To: [EMAIL PROTECTED]
   Subject: [Asterisk-Users] prepaid application
  
  
   Hello!
   I have installed the modified prepaid application and its working god.
 the
   only problem is that when I finish the call it does not update the
 balance
   of the card.
   any one has any idea how this could be fixed?
  
   best regards
   Hekuran
  
  
  
 
 
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Re: [Asterisk-Users] Patch for call queues?

[CW_ASN]

Please try CVS, AFAIK patch 214 doesn't included in stable branch.

Regards,

Gus

- Original Message - 
From: Robin Calmegård Siurua [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 30, 2004 7:10 PM
Subject: [Asterisk-Users] Patch for call queues?



 I'm looking for the patch that enables suppotr for the following lines
 in queue.conf:

 announce-holdtime = yes
 queue-youarenext = queue-youarenext
 queue-thereare = queue-thereare
 queue-callswaiting = queue-callswaiting
 queue-holdtime = queue-holdtime
 queue-minutes = queue-minutes
 queue-thankyou = queue-thankyou

 It doesn't work by default and I've lost the patch.. :/

 Would appreciate any help.


 /Robin - Swedish * newbie



 -- 
 Robin Calmegård Siurua
 CEO/developer
 RoCaS - development  support

 tel +46 8 505 556 80
 fax +46 8 505 556 79
 mobile +46 73 643 68 05
 [EMAIL PROTECTED]
 www.rocas.se

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Re: [Asterisk-Users] Midifyed-Prepaid-Application

[CW_ASN]

You have problems with pgsql. Check it.

Regards,

Gus

- Original Message - 
From: Hekuran Doli [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, June 20, 2004 5:27 PM
Subject: [Asterisk-Users] Midifyed-Prepaid-Application


 Hello.

 I have compile asterisk with modifyed prepaid application and populated
 the database to! I have fill the card, cardtype, cid, country,
 countrycode, reselers. I have make a cid=22 and I have add a user with
 username and callerid 22. But I allways get prepaid-no-aaa. Any one could
 help me how to authenticate?
 Note: I want to bill my local clients registred to my asterisk box using
sip.

 Best Regards
 Hekuran


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Re: [Asterisk-Users] Problems with PRI with T410 messages

[CW_ASN]

Send traces.


- Original Message - 
From: Aimable [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, June 17, 2004 6:28 AM
Subject: [Asterisk-Users] Problems with PRI with T410 messages


 Hi all,
 I have a box running asterisk with T410 connected to a Nortel DMS 100
switch
 and another box running SER with grandstream phones on it
 So if there is a call from the pstn it goes from the Nortel to the
asterisk
 and then to the SER box and finally to the phones.if the phone is busy or
 the number is invalid the * box will first send an ALERT message to the
 Nortel and say the call is going on and the phone is ringing (which is not
 the case )and after it will send a RELEASE  message saying that the line
is
 busy or the # is invalid .is there any way * can send a progress message
 instead of the alerting message until it gets the correct message from
SER?


 Thanks
 Habiyakare Aimable
 Phone Services
 TERRACOM Broadband
 [EMAIL PROTECTED]




 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED]
 Sent: Thursday, June 17, 2004 10:56 AM
 To: [EMAIL PROTECTED]
 Subject: Asterisk-Users digest, Vol 1 #4181 - 12 msgs

 Send Asterisk-Users mailing list submissions to
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 To subscribe or unsubscribe via the World Wide Web, visit
 http://lists.digium.com/mailman/listinfo/asterisk-users
 or, via email, send a message with subject or body 'help' to
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 When replying, please edit your Subject line so it is more specific
 than Re: Contents of Asterisk-Users digest...


 Today's Topics:

1. RE: Soekris Engineering net4801 (Senad Jordanovic)
2. Accepting SIP calls from unregistered gateways (Axel)
3. Re: pri with TE410P not working (Austria) (Peter Svensson)
4. Re: ZAPHFC - only for * 0.7.2? (Holger Schurig)
5. Calling the firefly network? (Martijn van Oosterhout)
6. RE: IAX2 no compatible codecs (Jason Penton)
7. Re: pri with TE410P not working (Austria) (Wolfgang Pichler)
8. Re: embedded Asterisk (Klaus-Peter Junghanns)
9. Re: pri with TE410P not working (Austria) (Michael Bielicki)
   10. RE: Cost of IP Phones, or Isn't It Just
Software? (Andy Powell)
   11. Re: pri with TE410P not working (Austria) (Peter Svensson)

 --__--__--

 Message: 1
 From: Senad Jordanovic [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Subject: RE: [Asterisk-Users] Soekris Engineering net4801
 Date: Thu, 17 Jun 2004 08:34:01 +0100
 Reply-To: [EMAIL PROTECTED]

 John Bittner wrote:
  Hi,
 
  I have it working great. I have debian running on it with music on
  hold disabled. I setup 10 cisco 7960 phones and tested the 4801 with
  calls on all 10 phones at the same time through voicepulse with no
  issues. I ran top with all the phones running and I was only up to
  45% cpu. Seems to run ok but I am still in the testing phase.

 Great...
 Have you tried to connect a X100P or TDM400P to it?


 --__--__--

 Message: 2
 From: Axel [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Date: Thu, 17 Jun 2004 03:43:12 -0400
 Subject: [Asterisk-Users] Accepting SIP calls from unregistered gateways
 Reply-To: [EMAIL PROTECTED]

 This is a multi-part message in MIME format.

 --=_NextPart_000_0351_01C4541D.36B45830
 Content-Type: text/plain;
 charset=iso-8859-1
 Content-Transfer-Encoding: quoted-printable

 Hi,
 Is there a way to accept SIP calls from unregistered gateways?
 autocreatpeer=3Dyes seems to disable checking credentials but the =
 originating gateway is still required to register itself with a username =
 and password (which can be anything since it won't check it).
 I like to be able to receive the call from any gateway without them =
 having to register even, just like a Cisco gateway that you can =
 terminate a call from clients who are not registered.  Is such thing =
 possible with Asterisk?

 Best regards,

 Axel

 --=_NextPart_000_0351_01C4541D.36B45830
 Content-Type: text/html;
 charset=iso-8859-1
 Content-Transfer-Encoding: quoted-printable

 !DOCTYPE HTML PUBLIC -//W3C//DTD HTML 4.0 Transitional//EN
 HTMLHEAD
 META http-equiv=3DContent-Type content=3Dtext/html; =
 charset=3Diso-8859-1
 META content=3DMSHTML 6.00.2800.1400 name=3DGENERATOR
 STYLE/STYLE
 /HEAD
 BODY bgColor=3D#ff
 DIVFONT face=3DArial size=3D2Hi,/FONT/DIV
 DIVFONT face=3DArial size=3D2Is there a way to accept SIP calls from =

 unregistered gateways?/FONT/DIV
 DIVFONT face=3DArial size=3D2autocreatpeer=3Dyes seems to disable =
 checking=20
 credentials but the originating gateway is still required to register =
 itself=20
 with a username and password (which can be anything since it won't check =

 it)./FONT/DIV
 DIVFONT face=3DArial size=3D2I like to be able to receive the call =
 from any=20
 gateway without them having to register even, just like a Cisco gateway =
 that you=20
 can terminate a call from clients who are not registered.nbsp; Is such =
 thing=20
 possible with Asterisk?/FONT/DIV

Re: [Asterisk-Users] Problems with PRI with T410 messages

[CW_ASN]

 This is a problem I pointed out to Digium a while back, but I am not sure
Markster understood the issue and I didn't really have the time to follow it
up.  It does need fixing though, as it is a major drawback in the current
architecture.

 Rgds
 Tim

 Hi all,
 I have a box running asterisk with T410 connected to a Nortel DMS 100
switch and another box running SER with grandstream phones on it So if there
is a call from the pstn it goes from the Nortel to the asterisk and then to
the SER box and finally to the phones.if the phone is busy or the number is
invalid the * box will first send an ALERT message to the Nortel and say the
call is going on and the phone is ringing (which is not the case )and after
it will send a RELEASE  message saying that the line is busy or the # is
invalid .is there any way * can send a progress message instead of the
alerting message until it gets the correct message from SER?


 Thanks
 Habiyakare Aimable


Call Proceeding can be sent only by transit network, not by the local switch
or pbx. AFAIK, * behavior for this scenario is like as local switch.
Certainly, this is not a normal behavior.

Regards,

Gus



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RE: [Asterisk-Users] Problems with PRI with T410 messages

[CW_ASN]

 
 I do not believe you are correct. We see CALL PROCEEDING in both
 directions as part of the normal ISDN call setup process.  See trace
 below.
 
 Asterisk sends 'CALL PROCEEDING' followed immediately by 'ALERTING'. CALL
 PROCEEDING is normally an acknowledgement to a SETUP. See Q931 below:
 
 3.1.2 CALL PROCEEDING
 This message is sent by the called user to the network or by the network
 to the calling user to indicate that requested call establishment has been
 initiated and no more call establishment information will be accepted. See
 Table 3-3.
 
 
 ALERTING has a very specific meaning:
 3.2.1 ALERTING
 This message is sent by the called user to the network to indicate that
 called user alerting has been initiated. See Table 3 23.
 
 i.e. the channel to the called party has been established, and the phone
 at the other end is physically ringing or making some other indication
 that an incoming call is there to be answered.
 
 It is 'ALERTING' that is being sent in the wrong place, as Asterisk sends
 'ALERTING' before the remote party (be it a SIP or IAX channel) is
 actually 'ringing'.  Receipt of 'ALERTING' from the called party is the
 trigger for the calling party to be presented with 'ringback tone'.  So to
 send a 'RELEASE' message with 'busy' after the caller has been told the
 phone is ringing is not a logical thing to do, and causes a lot of
 problems here.
 
 It needs fixing
 
 Rgds
 Tim

Tim:

Call proceeding is not mandatory in local termination (at least in
EuroISDN). Alerting is mandatory (obviously). Some class 5 switches sends
Call Proceeding only when the received SETUP will be routed thru CCS or CAS
routes, and only when a timer (I can't remember the timer number) expires.
The Call Proceeding must be retransmitted to A side. Call Proceeding message
is used mostly in transit environments.
Obviously, Ringing can't be used when unallocated or busy conditions are
detected.

The correct procedure for successful call with Call Proceeding and Setup
Acknowledge:
1) A-Setup
2) Setup acknowledge -B
3) Call Proceeding -B
4) Ringing -B
5) Answer -B
Or
5) Release A-B (by expiration time)

The correct procedure for unsuccessful (1 or 17 cause) call without Call
Proceeding, with Setup Acknowledge:
1) A-Setup
2) Setup acknowledge -B
3) Release -B (ITU-T release cause i.e.: 1 or 17)


As you said, it needs to be fixed.

Regards,

Gus



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Re: [Asterisk-Users] Prepaid application error

[CW_ASN]

Or compile the .so with -lpq option.



- Original Message - 

  From: 
  [EMAIL PROTECTED] 
  To: [EMAIL PROTECTED] 
  
  Sent: Tuesday, June 15, 2004 5:06 
AM
  Subject: Re: [Asterisk-Users] Prepaid 
  application error
  Hi, you have to launch the 
  script prepaid-make.sh in the database directory and copy the prepaid.conf in 
  /etc/asterisk. 
  


  reseaux [EMAIL PROTECTED] 
Sent by: [EMAIL PROTECTED] 

14/06/2004 18.24 

  
  

  Please respond 
  to[EMAIL PROTECTED]
  

  
  

  To
[EMAIL PROTECTED] 

  

  cc

  

  Subject
Re: [Asterisk-Users] 
  Prepaid application error

  
  

Dear list   
   I have try to compile app_prepaid with no problem but when i start * 
  (cvs branch) i have this kind of error:undefined symbol: PQexecCan 
  someone give some hits?Thanks in advanceDimitriOn Monday 14 
  June 2004 02:58 pm, [EMAIL PROTECTED] wrote: Hi, I 
  successfully installed postgres and prepaid application in my asterisk 
  box but after I digited the code I receive this error: ERROR: 
  Function asterisk_authenticate("unknown", "unknown") does not 
  exist Unable to identify a function that 
  satisfies the given argument types 
  You may need to add explicit typecasts   -- Playing 
  'prepaid-no-aaa' What is wrong ? 
  Bye___Asterisk-Users 
  mailing 
  list[EMAIL PROTECTED]http://lists.digium.com/mailman/listinfo/asterisk-usersTo 
  UNSUBSCRIBE or update options visit: 
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Re: [Asterisk-Users] Re: NetworkWorld article on Open Source Telephony

[CW_ASN]

Obviously, you have seen very few OM interfaces.

Regards,

Gus

- Original Message - 
From: W. Kevin Hunt [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, June 09, 2004 6:26 PM
Subject: RE: [Asterisk-Users] Re: NetworkWorld article on Open Source
Telephony


 I happen to feel that Cisco IOS is the most beautifull inteface known to
 present day man...

 W. Kevin Hunt
 CCIE #11841
 www.huntbrothers.com

 -Original Message-

 Subject: [Asterisk-Users] Re: NetworkWorld article on Open Source
 Telephony

  The power of asterisk comes from its method of config.

 yup.  it meets the challenge of finding something more complex, less
 intuitive, less parsable, and less managable than crisco ios.

 randy

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Re: [Asterisk-Users] Asterisk addons

[CW_ASN]

 - Original Message - 
 From: Fabio Donaggio
 To: [EMAIL PROTECTED]
 Sent: Friday, May 28, 2004 6:16 AM
 Subject: [Asterisk-Users] Asterisk addons


 Hi to all!!

 Is there another method to download asterisk addons???

 Thanks
 F

Man! Try to investigate for yourself! Use google!

http://www.google.com/search?q=asterisk-addons+downloadie=UTF-8hl=esmeta
=


Gus



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Re: [Asterisk-Users] Fw: Asterisk and MySQL

[CW_ASN]

 - Original Message - 
 From: Fabio Donaggio
 To: [EMAIL PROTECTED]
 Sent: Friday, May 28, 2004 12:52 PM
 Subject: [Asterisk-Users] Fw: Asterisk and MySQL


 Hi!

 It's all ok with CVS login...I download asterisk-addons.
 I would try to store sip friends in MySQL database and also the
voicemailcan you help me???
 Thanks


Again, use google and wiki...

http://www.google.com.ar/search?q=asterisk+sip+mysqlie=UTF-8hl=esmeta=
http://www.voip-info.org/wiki-Asterisk+sip+mysql+peers





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Re: [Asterisk-Users] X100P problem with PSTN from BOLIVIA

[CW_ASN]

Paste your extensions.conf
Check the answer command if you're running IVR of special services.


- Original Message -
From: Jorge Verastegui
To: [EMAIL PROTECTED]
Sent: Monday, May 17, 2004 8:46 PM
Subject: Re: [Asterisk-Users] X100P problem with PSTN from BOLIVIA


When i make a call from Asterisk everything goes Ok,
I do have a problem: when a call from the PSTN originates, the extension
in Asterisk hangs up and I only hear silence in the PSTN for
approximately 60 seconds.


On Mon, 2004-05-17 at 16:56, Juan J. Sierralta P. wrote:
 On Mon, 2004-05-17 at 12:10, Jorge Verastegui wrote:
  The silence last 60s (aprox)

 So maybe is the timeout used by your Telco (Entel?) Here at Chile we
 use 30s to let called people to be able to hang and get the call on
 another phone plugged to the same line.
 So I think it´s better to consult your telco.
 Does happen the same when the called party is a common phone not
 asterisk ?
--
Jorge Verastegui [EMAIL PROTECTED]
RedCetus S.R.L.





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RE: [Asterisk-Users] Webvmail

[CW_ASN]

make webvmail

from your source directory. Then, point your browser to:
http://your_ip/cgi-bin/vmail.cgi

Regards,

Gus

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Kurt
Enviado el: Miercoles, 21 de Abril de 2004 12:36 p.m.
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] Webvmail


I am having trouble locating webvmail on my * server.
Is this a seprate porgram or does it come with *.  I
am running version

asterick*CLI show version
Asterisk CVS-03/26/04-17:08:20 built by
[EMAIL PROTECTED] on a i686 running Linux
asterick*CLI


Thanks

Kurt




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RE: [Asterisk-Users] re: webvmail

[CW_ASN]

No, you don't need to change permissions. Check in your voicemail.conf the
user  password for accounts.
I don't know how vmail.cgi works with multiple contexts, or if you have
mysql/pgsql support with app_voicemail.

See http://www.voip-info.org/wiki-Asterisk+gui+vmail.cgi for more details.

Regards,

Gus

-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Kurt
Enviado el: Miercoles, 21 de Abril de 2004 02:05 p.m.
Para: [EMAIL PROTECTED]
Asunto: [Asterisk-Users] re: webvmail


Next question:

After doing your rerecommendation was able to get
to the main web page.  I trtriedogging in using one of
the vmvmailccounts (I am to assume that the login and
password is what I have set up in vovoicemailoconfor
mail boxes) and I got login incorrect.  Do i need to
change
permission on any of the files etc...

Kurt




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RE: [Asterisk-Users] Extention pickup

[CW_ASN]

http://www.voip-info.org/tiki-print.php?page=Asterisk+PBX+functions

  -Mensaje original-De: 
  [EMAIL PROTECTED] 
  [mailto:[EMAIL PROTECTED]En nombre de Kyle 
  HaganEnviado el: Martes, 20 de Abril de 2004 02:23 
  p.m.Para: [EMAIL PROTECTED]Asunto: 
  [Asterisk-Users] Extention pickup
   Does asterisk have a command 
  to pickup another ringing extention? I've tried searching but couldnt didnt 
  anything.
  
  Kyle
  

Re: [Asterisk-Users] Siemens EWSD 13

[CW_ASN]

In fact, with EWSD V13 you can't remove CRC4 in PRI mode.

Regards,

Gus

- Original Message -
From: Storer, Darren [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, April 07, 2004 8:32 PM
Subject: RE: [Asterisk-Users] Siemens EWSD 13


 Hi,

 I had exactly the same symptoms today with a co-located * connected to a
 Public Switch here in the UK. The problem was solved by insisting that the
 Telco turned on CRC4 at their end and then, after an 'init 6', layer two
 settled down on both systems.

 I was taught that if you are connecting to a full specification Q.931
 circuit, CRC4 should be enabled by default; in the event that one end does
 not support CRC4 the other end should auto-negotiate back and the circuit
 should still align without problems. Having said all of this I have yet to
 see auto-negotiation of CRC4 on any equipment (Public Network or CPE) and
 suspect that I was not told the truth in the first place...

 Selection of CRC4 seems to be random from Telco to Telco even on an
install
 by install basis within the same Carrier. It's the first thing to check
when
 new kit appears to be unstable..

 HTH

 Darren
 --
 Comgate
 TelcoInternetBroadcast

 -Original Message-
 From: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Behalf Of
 [EMAIL PROTECTED]
 Sent: 07 April 2004 14:59
 To: [EMAIL PROTECTED]
 Subject: [Asterisk-Users] Siemens EWSD 13


 Hi all,

 Has anyone got any experience with hooking Asterisk up with a
 Siemens EWSD 13 switch over a E1/PRI ?
 We're located in Belgium (Europe) and one of our telecom partners
 uses this switch.

 We connected one of our TE410P ports with their switch, but the status
 light on the TE410P card keeps blinking red.
 On their side they are getting a DSA (distance service alarm) error, so
 this normally means the devices 'see' eachother.. but there are still
 problems with the signalling.

 Our config below is the same as we are using for MCI, one of our other
 telecom partners.

 We tried changing the LBO and timing, but no luck.
 As you see the signalling is carried over channel 16 (default).

 TX and RX have also been regularly switched, so no luck..

 Their switch is providing the timing.

 The telecom operator has double checked the asterisk config several
 times, and it's conform to their setup.

 The only thing they couldn't find in the Asterisk config is a
 'multiframing' option. But I presume this is automatically detected or
 set by default ?
 They also tried normal/single(?) framing, but no difference.

 The card has also been tested with our MCI E1, and works flawlessly, so
 no hardware issue.

 Anyone got any further ideas ?

 Any info or help greatly appreciated!

 Our config,

 *** zaptel.conf ***
 span=1,1,6,ccs,hdb3,crc4,yellow
 bchan=1-15
 bchan=17-31
 dchan=16

 *** zapata.conf ***
 [channels]
 switchtype=euroisdn
 signalling=pri_cpe
 pridialplan=unknown

 group=1
 channel = 1-15,17-31

 other zapata standard config



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Re: [Asterisk-Users] Broken Asterisk

[CW_ASN]

When you see this message, try to kill mpg123 from another terminal (to stop
'Ouch...') and review the previous errors.

Regards,

Gus

- Original Message -
From: Simon Brown [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, March 28, 2004 10:37 PM
Subject: [Asterisk-Users] Broken Asterisk


 I don't know what I have done, but when I try to start Asterisk I get
 Ouch Error writing audio data: Broken pipe
 This scrolls endlessly and I cannot stop the screen except by killing the
 terminal session.

 TIA

 Simon

 -
 This mail was content checked for malicious code and viruses
 by GFI MailSecurity.

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Re: [Asterisk-Users] Basic authentication

[CW_ASN]

Try adding 'insecure=yes' in sip.conf.

Regards,

Gus

- Original Message -
From: Joao Carlos Moura [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, March 20, 2004 12:02 PM
Subject: [Asterisk-Users] Basic authentication


 How can I settup a way for Asterisk doesn´t make any use of  DIGEST
 AUTHENTICATION method?
 I don t want ASTERISK to check out any username or password of my users.


 Thank you

 Joao Carlos Moura

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Re: [Asterisk-Users] error, installing asterisk

[CW_ASN - Gus]

You can't expect much help without data...
Post the last compile messages, platform, SO.

Regards,

Gus

- Original Message - 
From: Hubert Kiyimba [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 15, 2004 5:31 AM
Subject: [Asterisk-Users] error, installing asterisk


 
 I got the following message while compiling asterisk file of the 
 asterisk-pbx 
 
 cannot find file lXI. 
 
 Please advise on what I should do. 
 
 hubert
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Re: [Asterisk-Users] (no subject)

[CW_ASN]

Alex:

In 'call' table stores call details.
'card' stores user  pin (10 digits in original version)
'country' associates a short description with a long description of
destination.
'countryprefix' associates prefix (i.e. 1305) with short description (of
'country' table) and type of destination (fixed, mobile).
'internationalprefix' contains a copy of countryprefix.prefix. Used in the
store procedure for match registers.
'providers' contains outgoing details:
'prefix' is the string added to the dialled string.
'providercode' is a code used to match rates, etc.
'providertech' is the protocol used. In the original version is not
posible to use zap devices. You must remove the ip in the dial command.
'providerip' is the ip for your gateway.
'providerdestination' contains the route codes for this provider.
'providerdestination.destination' is the same as countryprefix.prefix
'providerrate' stores the rate for each destination.
'providerrate.countrycode' is the same as short description for
'countryprefix' table.
'providerrate.subcode' is the same as type of destination in
'countryprefix' table.
'providerrate.rate' is the price (in cents) per minute.
'reseller','resselercard' and 'sale' unused at this time (I think).
'tariff' contains a tariff code and tariff name definitions.
'tariffrate' is almost the same as providerrate.

If your need more details or examples please advise.

Hope this helps.

Regards,

Gus


- Original Message -
From: Alexander Romanov [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 10, 2004 8:35 PM
Subject: [Asterisk-Users] (no subject)


 Hi guys,

 Has anyone played around/got it to work app_prepaid.c?
 (http://www.voip-info.org/wiki-Asterisk+callingcard)
 With what data do you populate the database with cards, providers,
 tariffs, tariffrates etc.. (format) to make it work. What is the
 meaning/purpose of each table/field?

 I am getting stuck here:

 Mar 11 10:33:28 DEBUG[1255670720]: app_prepaid.c:253
 prepaid_ivr_authorize: app_prepaid: SQL Authorize command as follows:
 SELECT * FROM asterisk_authorize('standard','61294332207') AS
 authorize(rate integer, tech text, prefix text, ipaddress text)
 Mar 11 10:33:28 DEBUG[1255670720]: rtp.c:950 ast_rtp_raw_write:
 Difference is 86856, ms is 10877
 Mar 11 10:33:28 DEBUG[1255670720]: channel.c:956 ast_settimeout:
 Scheduling timer at 160 sample intervals
 -- Playing 'prepaid-dest-unreachable' (language 'en')

 Thanks
 Alex.

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Re: [Asterisk-Users] SIP - Receptionist

[CW_ASN - Gus]

See monastery, maybe help you (http://pbx.unslept.com/newstatus.php)

Regards,

Gus

- Original Message - 
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 08, 2004 6:27 PM
Subject: [Asterisk-Users] SIP - Receptionist


 Hi All!
 I am thinking about fork-lift-upgrading a Nortel-Meridian
 key system with a * PBX driving SIP phones in the office. 
 The interface to PSTN would be a fractional T1 PRI (11 lines
 plus D channel). The GS phones look acceptable for most
 users. The forthcoming Sayson 480i would work for
 management types.  The receptionist, however, is currently
 used seeing a backlit display - with buttons - attached to
 her phone - showing all the extensions in the office, and
 who's has a conversation going etc.  We believe that
 autoattendant should only be used after hours ;).
 Question:  How do I drive - acquire such panels with
 asterisk? What are they called? who makes em?  I have seen
 Monastery, but that may be too cumbersome an interface for
 the relatively high call volume.
 I hope I explained what I am looking for.
 TIA 
 WW
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Re: [Asterisk-Users] SIP - Receptionist

[CW_ASN]

So put your hands on it and help to product grow.

Regards,

Gus

- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, March 08, 2004 8:19 PM
Subject: Re: [Asterisk-Users] SIP - Receptionist


 Monastery is neat as a monitoring tool.  The console's we're
 talking
 about also let the user pick-up calls etc.

 - Original Message Follows -
  See monastery, maybe help you
  (http://pbx.unslept.com/newstatus.php)
 
  Regards,
 
  Gus
 
  - Original Message -
  From: [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Monday, March 08, 2004 6:27 PM
  Subject: [Asterisk-Users] SIP - Receptionist
 
 
   Hi All!
   I am thinking about fork-lift-upgrading a
   Nortel-Meridian key system with a * PBX driving SIP
   phones in the office.  The interface to PSTN would be a
   fractional T1 PRI (11 lines plus D channel). The GS
   phones look acceptable for most users. The forthcoming
   Sayson 480i would work for management types.  The
   receptionist, however, is currently used seeing a
   backlit display - with buttons - attached to her phone -
   showing all the extensions in the office, and who's has
   a conversation going etc.  We believe that autoattendant
   should only be used after hours ;). Question:  How do I
   drive - acquire such panels with asterisk? What are they
   called? who makes em?  I have seen Monastery, but that
   may be too cumbersome an interface for the relatively
   high call volume. I hope I explained what I am looking
   for. TIA
   WW
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Re: [Asterisk-Users] Re: Limit on call in minuttes.

[CW_ASN]

You must change the setwhentohangup function, see channel.c for that.
Someone wrote a patch to do this (see http://bugs.digium.com/).

Regards,

Gus

- Original Message -
From: Hans-Henrik Andresen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, March 07, 2004 12:31 PM
Subject: [Asterisk-Users] Re: Limit on call in minuttes.



 Thank you This works, but. It just cut the line, I had hoped for some
 bip bip bip to remind that now your about to be disconected, is this
 possible as well ?

 /Hans-Henrik


 Senad Jordanovic [EMAIL PROTECTED] wrote in message
 news:[EMAIL PROTECTED]
  exten = 1,AbsoluteTimeout ($SECONDS)





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Re: [Asterisk-Users] Re: Re: Limit on call in minuttes.

[CW_ASN]

   This is wrongs. It's me who wrote the patch, it's available in CVS
Are you Klaus? If you're not Klaus, you wrote another patch. If you're
Klaus, as you see, works in that way.






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Re: [Asterisk-Users] Re: Re: Limit on call in minuttes.

[CW_ASN]

 Are you Klaus? If you're not Klaus, you wrote another patch. If you're
 Klaus, as you see, works in that way.
 
 
   Nopez i'm not

In that case, exists another patch from a guy called Klaus. I'm using this
patch since Dec2003.
Maybe helps, I don't know, but this is other alternative.
Its merged with the last app_dial from CVS, maybe isn't correct for the last
status (announce override).

Best regards,

Gus




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M/B 

[Asterisk-Users] Weird sdp output

[CW_ASN - Gus]

Hi all:

I'm doing some tests with sip equipments, and 
sometimes I see:

DEBUG[1150495040]: File chan_sip.c, Line 5077 
(handle_request): Hm No sdp for the moemnt
Does anyone knows anything about this?

Thanks in advance,

Gus

Re: [Asterisk-Users] Get new PRI working

[CW_ASN]

Why people don't have al least some respect about regulations?
Sure that pridial=unknown solved that problem, but sadly you're overwriting
the main class of service indication in ISDN...
Unknown let to Class 5 switch manage (as the operator wish) understand
your messages.
The common sense shows that the correct parameters maybe pridial=local,
where Class 5 switch don't add digits to the string.

The correct way to do this is calling to your operator, and ask for the
Class 5 brand and model (if the switch is Lucent, you need to use local.
With the rest of switches you can use all TON's).

Besides, the correct way to use PRI or S7 is to send ALWAYS the correct
Nature of address, not always the same...
In some parameter of your db you must define what prefix you use for
national calls and international calls.
The switch deletes the prefix when it was detected, and sends the correct
Nature Of Address for that call. This is a normal behavior for all kind of
switches. As far as I know, * always sends the same nature of address.

What's the difference between local and unknown? Local never add digits
and the calls will be treated mainly by the prefix that you send...
unknown was designed to try to match with any rule (really the first rule)
present in switch database.

Best regards,

Gus

- Original Message -
From: Tim Robinson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 15, 2004 12:16 PM
Subject: Re: [Asterisk-Users] Get new PRI working


 Adam -
 I had a similar problem here in the UK using a Euro-ISDN PRI from BT.
 The key was to add in the line pridialplan=unknown into zapata.conf.
 Then it leapt into life in both directions. My files are below for your
 information.

 Rgds
 Tim Robinson, Basingstoke UK


 zaptel.conf
 ---
 # Config for a UK Euro-ISDN line

 span=1,1,0,ccs,hdb3,crc4
 bchan=1-15
 dchan=16
 bchan=17-31
 loadzone=uk
 defaultzone=uk

 zapata.conf
 ---
 ; Configuration file

 [channels]
 usecallerid=yes
 language=en
 pridialplan=unknown
 signalling=pri_cpe
 switchtype=euroisdn
 group=1
 context=inboundpstn
 channel = 1-15
 channel = 17-31



 Adam Goryachev wrote:
  Hi all,
 
  I received my shiny new TE405P on Friday, and after much fiddling and
  assistance from the irc channel, I got a OK status (telco reversed the
TX/RX
  and I wired it wrong).
 
  Anyway, currently it works for inbound calls, but I can't seem to
dialout on
  it. Here is the config from zaptel.conf:
  span=1,1,0,ccs,hdb3,crc4
  bchan=1-10
  unused=11-15,17-31
  dchan=16
 
  and zapata.conf
  switchtype = euroisdn
  callgroup = 1
  group = 2
  busydetect = no
  immediate = yes
  context = remote
  signalling = pri_cpe
  ;stripmsd = 1
  callprogress = no
  channel = 1-10
 
  and here is the debug from asterisk:
  -- Executing Dial([EMAIL PROTECTED]:4569]/3,
Zap/2/93454395||rT)
  in new stack
  Feb 15 15:58:27 DEBUG[20497]: app_dial.c:400 dial_exec: SIMPLE DIAL (NO
URL)
  -- Making new call for cr 32774
 
 Protocol Discriminator: Q.931 (8)  len=43
 Call Ref: len= 2 (reference 6/0x6) (Originator)
 Message type: SETUP (5)
 Bearer Capability (len= 3) [ Ext: 1  Q.931 Std: 0  Info transfer
 
  capability: Speech (0)
 
  Ext: 1  Trans mode/rate: 64kbps,
circuit-mode
 
  (16)
 
  Ext: 1  User information layer 1: A-Law
(35)
 Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
 
  Dchan: 0
 
ChanSel: Reserved
   Ext: 1  Coding: 0   Number Specified   Channel
Type: 3
   Ext: 1  Channel: 2 ]
 Display (len= 7) [  Display (len= 7) [ 1 Display (len= 7) [ 1H
Display
 
  (len= 7) [ 1Ho Display (len= 7) [ 1Hom Display (len= 7) [ 1Home
Display
  (len= 7) [ 1Home  Display (len= 7) [ 1Home 2 Display (len= 7) [ 1Home
2 ]
 
 Calling Number (len= 7) [ Ext: 0  TON: National Number (2)  NPI:
 
  ISDN/Telephony Numbering Plan (E.164/E.163) (1)
 
   Presentation: Presentation permitted, user
 
  number passed network screening (1) '651' ]
 
 Called Number (len=11) [ Ext: 1  TON: National Number (2)  NPI:
 
  ISDN/Telephony Numbering Plan (E.164/E.163) (1) '93454395' ]
 
 Sending Complete (len= 0)
 
  -- Called 2/93454395
   Protocol Discriminator: Q.931 (8)  len=13
   Call Ref: len= 2 (reference 32774/0x8006) (Terminator)
   Message type: STATUS (125)
   Cause (len= 3) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location:
  Public network serving the local user (2)
Ext: 1  Cause: Info. element nonexist or not
implemented
  (99), class = Protocol Error (6) ]
Cause data 0: 01 (1)
   Call State (len= 1) [ Ext: 0  Coding: CCITT (ITU) standard (0) Call
state:
  Call Initiated (1)
  -- Processing IE 8 (Cause)
  -- Processing IE 20 (Call State)
   Protocol Discriminator: Q.931 (8)  len=10
   Call Ref: len= 2 (reference 32774/0x8006) (Terminator)
   Message type: CALL PROCEEDING (2)
   Channel ID (len= 5) [ Ext: 

Re: [Asterisk-Users] Problems with ATA's locking up..

[CW_ASN]

3.0.0 have some problems. Sometimes, ata answers to invite with Not found
or Busy here. This is a strange behavior.
I'm using now 2.16.2

Regards,

Gus


- Original Message -
From: Billy Huddleston [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 08, 2004 12:56 AM
Subject: [Asterisk-Users] Problems with ATA's locking up..


 Anyone had any problems with ATA's running 3.0 software locking up?

 Thanks, Billy

  +--+
  | Billy HuddlestonSenior System Administrator  |
  | Net-Express  http://www.nxs.net  |
  | 114 Sherway Rd. Voice: 865-691-2011  |
  | Knoxville, TN  37922  Fax: 865-691-9894  |
  | [EMAIL PROTECTED]|
  +--+
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Re: [Asterisk-Users] Problems with ATA's locking up..

[CW_ASN]

Could you share your 3.0.0 config?

- Original Message -
From: Florian Overkamp [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 08, 2004 2:10 PM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..


 Hi,

 Citeren CW_ASN [EMAIL PROTECTED]:

  3.0.0 have some problems. Sometimes, ata answers to invite with Not
found
  or Busy here. This is a strange behavior.
  I'm using now 2.16.2

 Hm ? I have not seen this happening yet. 2.16 has alternative behaviour
 regarding flash transfers...

 Florian
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Re: [Asterisk-Users] Problems with ATA's locking up..

[CW_ASN]

I will test with TOS in a8b8. All other stuff are equal in my ata.

Regards,

Gus

- Original Message -
From: Billy Huddleston [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, February 08, 2004 2:51 PM
Subject: Re: [Asterisk-Users] Problems with ATA's locking up..


 http://www.nxs.net/cisco_ata_186.htm


 - Original Message -
 From: CW_ASN [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Sunday, February 08, 2004 12:40 PM
 Subject: Re: [Asterisk-Users] Problems with ATA's locking up..


  Could you share your 3.0.0 config?
 
  - Original Message -
  From: Florian Overkamp [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Sunday, February 08, 2004 2:10 PM
  Subject: Re: [Asterisk-Users] Problems with ATA's locking up..
 
 
   Hi,
  
   Citeren CW_ASN [EMAIL PROTECTED]:
  
3.0.0 have some problems. Sometimes, ata answers to invite with Not
  found
or Busy here. This is a strange behavior.
I'm using now 2.16.2
  
   Hm ? I have not seen this happening yet. 2.16 has alternative
behaviour
   regarding flash transfers...
  
   Florian
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Re: [Asterisk-Users] question for oh323 users

[CW_ASN - Gus]

It must be:

exten = _1905XXX,1,Dial,OH323/h323:[EMAIL PROTECTED]
or
exten = _1905XXX,1,Dial,OH323/[EMAIL PROTECTED]

Hope this helps,

Gus

- Original Message -
From: Anthony Law [EMAIL PROTECTED]
To: Mailing List Asterisk [EMAIL PROTECTED]
Sent: Friday, February 06, 2004 11:56 AM
Subject: Re: [Asterisk-Users] question for oh323 users


 Hi,

 Thanks for your reply. I am definite that my h323 is running on ciscoB
 because the below scenario is working fine.

 pstnciscoA-ciscoBpstn

 I have also eliminated access-list problem because if my access-list is
 applied I could see packets hiting my access-list

 permit tcp host 192.168.1.2 any eq 1720 (60 matches)

 Is my syntax below correct ??

 exten = _1905XXX,1,Dial,OH323/192.168.1.3

 Any help would be appreciated.


 Regards,



 Anthony


 - Original Message -
 From: Tomica Crnek [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Friday, February 06, 2004 3:03 AM
 Subject: RE: [Asterisk-Users] question for oh323 users


 
  Hi, it seams to me that h.323 service on your cisco B could be down. You
  see packets coming to this box, but did you activate h.323. Try telnet
  192.168.1.3 1720 to see if it is running. If it is, then check to see
  if you are allowing connections to it from 192.168.1.2
 
  Tomica
 
 
  -Original Message-
  From: [EMAIL PROTECTED]
  [mailto:[EMAIL PROTECTED] On Behalf Of Anthony Law
  Sent: Thursday, February 05, 2004 10:41 PM
  To: [EMAIL PROTECTED]
  Subject: [Asterisk-Users] question for oh323 users
 
  Hi,
 
  I am trying to forward calls from one cisco gateway to another cisco
  gateway using asterisk
 
  cisco(5300)A 192.168.1.1
  asterisk 192.168.1.2
  cisco(5300)B 192.168.1.3
 
  pstn --ciscoA-asterisk --ciscoB--pstn
 
  I have the below in my extension.conf
 
  [default]
  exten = _1905XXX,1,Dial,OH323/192.168.1.3
 
  I keep getting error and I don't know what is wrong.
  I am able to see in my ciscoB accesslist, tcp packets are coming from
  192.168.1.2
 
  I get below error in my asterisk CLI
 
  Feb  5 16:17:01 ERROR[29716]: chan_oh323.c:1004 oh323_call: H323:0:
  Could not call 192.168.1.3.
  Feb  5 16:17:11 WARNING[29716]: pbx.c:1829 ast_pbx_run: Timeout, but no
  rule 't' in context 'default'
 
  It would be much appreciated if someone could point out what I am doing
  wrong or to any documentations. Many thanks.
 
 
  Regards,
 
 
 
  Anthony
 
 
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Re: [Asterisk-Users] Meetme without zaptel hardware

[CW_ASN - Gus]

Yes, lot of people use ztdummy.


  - Original Message - 
  From: 
  Paul 
  To: [EMAIL PROTECTED] 
  
  Sent: Monday, February 02, 2004 12:49 
  AM
  Subject: [Asterisk-Users] Meetme without 
  zaptel hardware
  
  
  Has anyone had any success using 
  the ztdummy module and doing meetme/conferencing with out zaptel hardware 
  installed?
  
  Paul

Re: [Asterisk-Users] HANGUPCAUSE

[CW_ASN - Gus]

How? Is written in CDR?

Regards,

Gus

- Original Message - 
From: Tais M. Hansen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 9:20 AM
Subject: Re: [Asterisk-Users] Echo worsens in 0.7.1


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Friday 30 January 2004 00:57, Eric Wieling wrote:
 Is there any chance 0.7.2 will include a fix for PRI Cause Codes not
 being translated into Asterisk Cause Codes and being passed back to
 app_dial (as well as fixing the apparently never working ${HANGUPCAUSE}
 variable)?

HANGUPCAUSE is working fine here (cvs).

- -- 
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Version: GnuPG v1.2.3 (GNU/Linux)

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pRRyhh0J/GeyezwX1m8Qi1s=
=PbAl
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Re: [Asterisk-Users] HANGUPCAUSE

[CW_ASN - Gus]

Ok, but is not working as expected... we can't see clear ISUP causes. We
can't make different treatments or store other causes than busy (cause=17)
in cdr's .

Regards,

Gus

- Original Message -
From: Tais M. Hansen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 9:48 AM
Subject: Re: [Asterisk-Users] HANGUPCAUSE


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Friday 30 January 2004 13:31, CW_ASN - Gus wrote:
 HANGUPCAUSE is working fine here (cvs).
 How? Is written in CDR?

CDRs contain BUSY when busy and NO ANSWER on the rest.

extensions.conf:

[provider-out]
...
exten = _XX.,7,Dial(ZAP/g1/${calledid}|120|r)
exten = _XX.,8,Goto(provider-out-failed|c${HANGUPCAUSE}|1)

[provider-out-failed]
exten = c1,1,Hangup()

exten = c2,1,Busy()

exten = c3,1,Answer()
exten = c3,2,ResetCDR()
exten = c3,3,Playtones(info)
exten = c3,4,Wait(60)
exten = c3,5,Hangup()

exten = c4,1,Congestion()

- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Version: GnuPG v1.2.3 (GNU/Linux)

iD8DBQFAGlKy2TEAILET3McRAv7gAKCREpAN3kVvbEuTDAQkU9kb6IrZiQCdEXlR
3FroTgPgWQmBrqGwjwktmvc=
=yyxo
-END PGP SIGNATURE-

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Re: [Asterisk-Users] HANGUPCAUSE

[CW_ASN - Gus]

It would to be good in any way... :)


- Original Message -
From: Tais M. Hansen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 30, 2004 12:57 PM
Subject: Re: [Asterisk-Users] HANGUPCAUSE


-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1

On Friday 30 January 2004 15:59, CW_ASN - Gus wrote:
 Ok, but is not working as expected... we can't see clear ISUP causes. We
 can't make different treatments or store other causes than busy (cause=17)
 in cdr's .

You could use my approach and combine it with the CDR userfield. Personally
I
would like a PRI_CAUSE variable to be set as well as HANGUPCAUSE.

- --
Regards,
Tais M. Hansen
ComX Networks
Tel: +45-70257474
Fax: +45-70257374
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Version: GnuPG v1.2.3 (GNU/Linux)

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5B+arXbMx37BtKSFLez3KlI=
=61o0
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Re: [Asterisk-Users] app_queue and dialplan

[CW_ASN - Gus]

Try with:

http://bugs.digium.com/bug_view_page.php?bug_id=214

Regards,

Gus


- Original Message - 
From: Anton Yurchenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26, 2004 11:01 AM
Subject: [Asterisk-Users] app_queue and dialplan


 Hello,
 
 I`m trying to achive this:
 1. when the initial call comes in it is served by a small queue with 
 short timeout so that at first caller hears only ringing
 2. if nobody answers the call at that time or the queue is all full the 
 call goes to the Playback the message ( please hold bla bla bla)
 3. Then the call goes to another queue and he holds while the 
 music-on-hold plays a app_queue trys to reach the next free operator
 4. after a timeout in second queue there is a Goto to play the message 
 again and then back into the second queue
 
 
 I have it like this:
 
 extensions.conf:
 exten = 10,1,Queue(q1_short,tn)
 exten = 10,2,Answer
 exten = 10,3,Playback(please_hold)
 exten = 10,4,Queue(q1,t)
 exten = 10,5,Goto(3)
 
 
 queue.conf:
 
 [q1]
 music = test
 announce = test_anounce
 timeout = 40
 retry = 3
 maxlen = 10
 
 strategy = leastrecent
 
 member = SIP/111
 member = SIP/112
 member = SIP/113
 member = SIP/114
 member = SIP/115
 
 
 [q1_short]
 music = test
 announce = test_anounce
 timeout = 15
 retry = 3
 maxlen = 3
 strategy = leastrecent
 member = SIP/111
 member = SIP/112
 member = SIP/113
 member = SIP/114
 member = SIP/115
 
 
 but the broblem is when the q1_short is full, and the call goes to the 
 q1 it only plays the announce message and and no music on hold is played 
 and again the  announce message is played. somehow the music on lod 
 doesn start. What am I doing wrong?
 I run version CVS-12/01/03-14:50:57
 
 Thanks
 
 
 -- 
 
 Anton Yurchenko[EMAIL PROTECTED]
 Digital Generation
 
 
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Re: [Asterisk-Users] app_queue and dialplan

[CW_ASN - Gus]

Try with:

http://bugs.digium.com/bug_view_page.php?bug_id=214

Regards,

Gus

- Original Message - 
From: Anton Yurchenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 26, 2004 9:59 AM
Subject: [Asterisk-Users] app_queue and dialplan


 Hello,
 
 I`m trying to achive this:
 1. when the initial call comes in it is served by a small queue with 
 short timeout so that at first caller hears only ringing
 2. if nobody answers the call at that time or the queue is all full the 
 call goes to the Playback the message ( please hold bla bla bla)
 3. Then the call goes to another queue and he holds while the 
 music-on-hold plays a app_queue trys to reach the next free operator
 4. after a timeout in second queue there is a Goto to play the message 
 again and then back into the second queue
 
 
 I have it like this:
 
 extensions.conf:
 exten = 10,1,Queue(q1_short,tn)
 exten = 10,2,Answer
 exten = 10,3,Playback(please_hold)
 exten = 10,4,Queue(q1,t)
 exten = 10,5,Goto(3)
 
 
 queue.conf:
 
 [q1]
 music = test
 announce = test_anounce
 timeout = 40
 retry = 3
 maxlen = 10
 
 strategy = leastrecent
 
 member = SIP/111
 member = SIP/112
 member = SIP/113
 member = SIP/114
 member = SIP/115
 
 
 [q1_short]
 music = test
 announce = test_anounce
 timeout = 15
 retry = 3
 maxlen = 3
 strategy = leastrecent
 member = SIP/111
 member = SIP/112
 member = SIP/113
 member = SIP/114
 member = SIP/115
 
 
 but the broblem is when the q1_short is full, and the call goes to the 
 q1 it only plays the announce message and and no music on hold is played 
 and again the  announce message is played. somehow the music on lod 
 doesn start. What am I doing wrong?
 I run version CVS-12/01/03-14:50:57
 
 Thanks
 
 
 -- 
 
 Anton Yurchenko[EMAIL PROTECTED]
 Digital Generation
 
 
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Re: [Asterisk-Users] G.723.1

[CW_ASN]

If you don't have the licences for this codec, you can't playback files from
*.
If I'm not mistaken, * can be used to do codec passthrough between two
endpoints, but you can't use any application to interact with *, like
voicemail, directory, background or playback.

Regards,

Gus


- Original Message -
From: Cesar Rico
To: [EMAIL PROTECTED]
Sent: Friday, January 23, 2004 7:03 PM
Subject: [Asterisk-Users] G.723.1



Hi all,

I have a g.723.1 file and my voice devices support this codec, I need to
playback this file in asterisk , I stored it in the directory
/var/lib/asterisk/sounds/ but when I executte the command in the
extension.conf (exten = 100,1,playback(file.g7323) the call hang up, my
voice devices are configured with g723 codec, I read that * pass through
this codec, so I don't know why this configuration don't work well, if
anybody have some idea to respet let me know.

I will appreciate you support

Best regards

Cesar Rico.




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Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

[CW_ASN - Gus]

The incoming call request Unrestricted and 64K, and this looks like ok, but
in the SETUP_ACK the called number parameters shows: Ext: 1  Progress
Description: Called equipment is non-ISDN. (2) ], like as is not an ISDN
equipment.
In the most of cases, Information transfer rate = to '64 kbit/s', and Info
transfer capability = 'real bw required'.

Are you sure that the equipment attached to * can be used in 64K?

Regards,

Gus

- Original Message -
From: Thomas Haeger [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 12:28 PM
Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


 Hi ,

 maybe someone knows what's going wrong...

 The incoming data call will not really identified as ISDN 64k/Data

 Here my pri debug ouput

  Protocol Discriminator: Q.931 (8)  len=39
  Call Ref: len= 2 (reference 5635/0x1603) (Originator)
  Message type: SETUP (5)
  Bearer Capability (len= 2) [ Ext: 1  Q.931 Std: 0  Info transfer
 capability: Unrestricted digital information (8)
   Ext: 1  Trans mode/rate: 64kbps,
circuit-mode
 (16)
   Ext: 0  User information layer 1: Unknown
 (24)
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred
 Dchan: 0
 ChanSel: Reserved
Ext: 1  Coding: 0   Number Specified   Channel
Type:
 3
Ext: 1  Channel: 30 ]
  Calling Number (len=14) [ Ext: 0  TON: National Number (2)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
 number not screened (0) '3328334778' ]
  Called Number (len=11) [ Ext: 1  TON: Unknown Number Type (0)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1) '63494441' ]
 -- Making new call for cr 5635
 -- Processing Q.931 Call Setup
 -- Processing IE 4 (Bearer Capability)
 -- Processing IE 24 (Channel Identification)
 -- Processing IE 108 (Calling Party Number)
 -- Processing IE 112 (Called Party Number)
  Protocol Discriminator: Q.931 (8)  len=14
  Call Ref: len= 2 (reference 38403/0x9603) (Terminator)
  Message type: SETUP ACKNOWLEDGE (13)
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
 Dchan: 0
 ChanSel: Reserved
Ext: 1  Coding: 0   Number Specified   Channel
Type:
 3
Ext: 1  Channel: 30 ]
  Progress Indicator (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0)
0:
 0   Location: Private network serving the local user (1)
Ext: 1  Progress Description: Called
 equipment is non-ISDN. (2) ]
 -- Accepting call from '3328334778' to '63494441' on channel 30, span
2
 -- Executing GotoIf(Zap/61-1, 0?50:100) in new stack
 -- Goto (pri2,63494441,100)
 -- Executing Dial(Zap/61-1, Zap/g2/033283077733SPEECH) in new
stack
 -- Making new call for cr 39439
  Protocol Discriminator: Q.931 (8)  len=50
  Call Ref: len= 2 (reference 6671/0x1A0F) (Originator)
  Message type: SETUP (5)
  Bearer Capability (len= 3) [ Ext: 1  Q.931 Std: 0  Info transfer
 capability: Speech (0)
   Ext: 1  Trans mode/rate: 64kbps,
circuit-mode
 (16)
   Ext: 1  User information layer 1: A-Law
(35)
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred
 Dchan: 0
 ChanSel: Reserved
Ext: 1  Coding: 0   Number Specified   Channel
Type:
 3
Ext: 1  Channel: 1 ]
  Calling Number (len=14) [ Ext: 0  TON: Subscriber Number (4)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user
 number passed network screening (1) '3328334778' ]
  Called Number (len=21) [ Ext: 1  TON: Subscriber Number (4)  NPI:
 ISDN/Telephony Numbering Plan (E.164/E.163) (1) '033283077733SPEECH' ]
 -- Called g2/033283077733SPEECH
  Protocol Discriminator: Q.931 (8)  len=10
  Call Ref: len= 2 (reference 39439/0x9A0F) (Terminator)

  Message type: SETUP ACKNOWLEDGE (13)
  Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Exclusive
 Dchan: 0
 ChanSel: Reserved
Ext: 1  Coding: 0   Number Specified   Channel
Type:
 3
Ext: 1  Channel: 1 ]
 -- Processing IE 24 (Channel Identification)
 beroasterisk*CLI
  Protocol Discriminator: Q.931 (8)  len=9
  Call Ref: len= 2 (reference 5635/0x1603) (Originator)
  Message type: DISCONNECT (69)
  Cause (len= 2) [ Ext: 1  Coding: CCITT (ITU) standard (0) 0: 0
Location:
 User (0)
   Ext: 1  Cause: Normal Clearing (16), class = Normal
Event
 (1) ]
 -- Processing IE 8 (Cause)
 -- Channel 30, span 2 got hangup
 NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Overlap sending, peerstate
 Overlap Receiving
  Protocol Discriminator: Q.931 (8)  len=9
  Call Ref: len= 2 (reference 6671/0x1A0F) (Originator)
  

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

[CW_ASN - Gus]

You're still receiving [Incompatible destination], this cause is used when
bearer capabilities aren't equal.


- Original Message -
From: Thomas Haeger [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 3:07 PM
Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


 Hi,

 we tried following scenario:


 DTAG (S0) at our office Datacall with AVMFritz (PSTN) --- Colo
 TelesSwitch -- CoLo Asterisk (--- PSTN)

 I think, no i know that the Teles Switch can route 64k data calls
 here is the Teles Trace:

 #08SETUP--|
 15:29:40,378 02 01 78 AE   |
  08 02 03 90 05|
 Bearer Caps  04 02 88 90   |
 Channel Id   18 03 A1 83 9B|
 Calling PN   6C 0C 21 83 33 33 32 38   |
  33 33 34 37 37 38 |
 Called PN70 09 C1 36 33 34 39 34   |
  34 34 31  |
|--RR
 #08
|   15:29:40,388 02 01 01 7A
|--SETUP ACKNOWLEDGE
 #08
|   15:29:40,398 00 01 AE 7A
|08 02 83 90 0D
|   Channel Id   18 03 A9 83 9B
|--SETUP
 #12
|   15:29:40,408 00 01 2A D4
|08 02 16 60 05
|   Bearer Caps  04 02 88 90
|   Channel Id   18 03 A1 83 88
|   Calling PN   6C 0C 21 80 33 33
32
 38
|33 33 34 37 37 38
|   Called PN70 09 81 36 33 34
39
 34
|34 34 31
 #08   RR--|
 15:29:40,408 00 01 01 B0   |
 #12   RR--|
 15:29:40,418 00 01 01 2C   |
 #12SETUP ACKNOWLEDGE--|
 15:29:40,418 02 01 D4 2C   |
  08 02 96 60 0D|
 Channel Id   18 03 A9 83 88|
 Progress Ind 1E 02 81 82   |
|--RR
 #12
|   15:29:40,418 02 01 01 D6
 #12SETUP--|
 15:29:40,428 02 01 D6 2C   |
  08 02 1A 21 05|
 Bearer Caps  04 03 88 90 A3|
 Channel Id   18 03 A1 83 81|
 Calling PN   6C 0C 41 81 33 33 32 38   |
  33 33 34 37 37 38 |
 Called PN70 0D C1 30 33 33 32 38   |
  33 30 37 37 37 33 33  |
|--RELEASE COMPLETE
 #12
|   15:29:40,428 00 01 2C D8
|08 02 9A 21 5A
|08 02 80 D8
|[Incompatible
 destinat
|ion]
 #12   RR--|

 -Ursprüngliche Nachricht-
 Von: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Auftrag von CW_ASN -
 Gus
 Gesendet: Donnerstag, 22. Januar 2004 17:24
 An: [EMAIL PROTECTED]
 Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


 The incoming call request Unrestricted and 64K, and this looks like ok,
but
 in the SETUP_ACK the called number parameters shows: Ext: 1  Progress
 Description: Called equipment is non-ISDN. (2) ], like as is not an ISDN
 equipment.
 In the most of cases, Information transfer rate = to '64 kbit/s', and Info
 transfer capability = 'real bw required'.

 Are you sure that the equipment attached to * can be used in 64K?

 Regards,

 Gus

 - Original Message -
 From: Thomas Haeger [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Thursday, January 22, 2004 12:28 PM
 Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


  Hi ,
 
  maybe someone knows what's going wrong...
 
  The incoming data call will not really identified as ISDN 64k/Data
 
  Here my pri debug ouput
 
   Protocol Discriminator: Q.931 (8)  len=39
   Call Ref: len= 2 (reference 5635/0x1603) (Originator)
   Message type: SETUP (5)
   Bearer Capability (len= 2) [ Ext: 1  Q.931 Std: 0  Info transfer
  capability: Unrestricted digital information (8)
Ext: 1  Trans mode/rate: 64kbps,
 circuit-mode
  (16)
Ext: 0  User information layer 1: Unknown
  (24)
   Channel ID (len= 5) [ Ext: 1  IntID: Implicit, PRI Spare: 0, Preferred
  Dchan: 0
  ChanSel: Reserved
 Ext: 1  Coding: 0   Number Specified   Channel
 Type:
  3

Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI

[CW_ASN - Gus]

I don't know, but I can test in a very short time.
I let you know for details.

- Original Message -
From: Thomas Haeger [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 4:07 PM
Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


 Has somebody got it work at all ?
 I mean data calls (ISDN 64k) through asterisk.

 Regards,

 Thomas.

 -Ursprüngliche Nachricht-
 Von: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Auftrag von Thomas
 Haeger
 Gesendet: Donnerstag, 22. Januar 2004 19:07
 An: [EMAIL PROTECTED]
 Betreff: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


 Hi,

 we tried following scenario:


 DTAG (S0) at our office Datacall with AVMFritz (PSTN) --- Colo
 TelesSwitch -- CoLo Asterisk (--- PSTN)

 I think, no i know that the Teles Switch can route 64k data calls
 here is the Teles Trace:

 #08SETUP--|
 15:29:40,378 02 01 78 AE   |
  08 02 03 90 05|
 Bearer Caps  04 02 88 90   |
 Channel Id   18 03 A1 83 9B|
 Calling PN   6C 0C 21 83 33 33 32 38   |
  33 33 34 37 37 38 |
 Called PN70 09 C1 36 33 34 39 34   |
  34 34 31  |
|--RR
 #08
|   15:29:40,388 02 01 01 7A
|--SETUP ACKNOWLEDGE
 #08
|   15:29:40,398 00 01 AE 7A
|08 02 83 90 0D
|   Channel Id   18 03 A9 83 9B
|--SETUP
 #12
|   15:29:40,408 00 01 2A D4
|08 02 16 60 05
|   Bearer Caps  04 02 88 90
|   Channel Id   18 03 A1 83 88
|   Calling PN   6C 0C 21 80 33 33
32
 38
|33 33 34 37 37 38
|   Called PN70 09 81 36 33 34
39
 34
|34 34 31
 #08   RR--|
 15:29:40,408 00 01 01 B0   |
 #12   RR--|
 15:29:40,418 00 01 01 2C   |
 #12SETUP ACKNOWLEDGE--|
 15:29:40,418 02 01 D4 2C   |
  08 02 96 60 0D|
 Channel Id   18 03 A9 83 88|
 Progress Ind 1E 02 81 82   |
|--RR
 #12
|   15:29:40,418 02 01 01 D6
 #12SETUP--|
 15:29:40,428 02 01 D6 2C   |
  08 02 1A 21 05|
 Bearer Caps  04 03 88 90 A3|
 Channel Id   18 03 A1 83 81|
 Calling PN   6C 0C 41 81 33 33 32 38   |
  33 33 34 37 37 38 |
 Called PN70 0D C1 30 33 33 32 38   |
  33 30 37 37 37 33 33  |
|--RELEASE COMPLETE
 #12
|   15:29:40,428 00 01 2C D8
|08 02 9A 21 5A
|08 02 80 D8
|[Incompatible
 destinat
|ion]
 #12   RR--|

 -Ursprüngliche Nachricht-
 Von: [EMAIL PROTECTED]
 [mailto:[EMAIL PROTECTED] Auftrag von CW_ASN -
 Gus
 Gesendet: Donnerstag, 22. Januar 2004 17:24
 An: [EMAIL PROTECTED]
 Betreff: Re: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


 The incoming call request Unrestricted and 64K, and this looks like ok,
but
 in the SETUP_ACK the called number parameters shows: Ext: 1  Progress
 Description: Called equipment is non-ISDN. (2) ], like as is not an ISDN
 equipment.
 In the most of cases, Information transfer rate = to '64 kbit/s', and Info
 transfer capability = 'real bw required'.

 Are you sure that the equipment attached to * can be used in 64K?

 Regards,

 Gus

 - Original Message -
 From: Thomas Haeger [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Thursday, January 22, 2004 12:28 PM
 Subject: AW: [Asterisk-Users] Data calls (ISDN/64k) through * PRI


  Hi ,
 
  maybe someone knows what's going wrong...
 
  The incoming data call will not really identified as ISDN 64k/Data
 
  Here my pri debug ouput
 
   Protocol Discriminator: Q.931 (8)  len=39
   Call Ref: len= 2 (reference 5635/0x1603) (Originator)
   Message type: SETUP (5)
   Bearer Capability (len= 2) [ Ext: 1  Q.931 Std: 0  Info transfer
  capability: Unrestricted digital information (8)
Ext: 1  Trans mode/rate

Re: [Asterisk-Users] ETSI PRI ISDN Signalling

[CW_ASN - Gus]

Please send your zaptel.conf to see what's going on.


- Original Message - 
From: Daniel Bichara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, January 22, 2004 4:38 PM
Subject: [Asterisk-Users] ETSI PRI ISDN Signalling


 Hi All,
 
 I've bought a R2Adapter to convert R2Digital to ISDN. I am trying to 
 config E100P card but D-Channel is down. I know R2Adapter uses ETSI ISDN 
 Protocol so I tried unsucessfully setup switchtype to EuroISDN. Any clue?
 
 Daniel
 
 
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Re: [Asterisk-Users] R2 support

[CW_ASN - Gus]

 Maybe Telefonica (the same from .ar) is not big enough!

By the sight Telefónica in Brazil is not very serious, in Argentina offers
ISDN in all country, for all kinds of teleservices... I'm sure of that.


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Re: [Asterisk-Users] R2 support

[CW_ASN]

CW_ASN - Gus wrote:

  
Ok, it's old and clunky, but in some countries like Brazil, Argentina and
China is the only alternative.
Only alternative??? Why is the only alternative? All mayor carriers in
Argentina and Brasil have PRI signalling, at the same price.

  
  Sorry, but you are wrong. I am from Brazil and E1-ISDN is not avaible all 
  over the country.Daniel
  Maybe, you don't have big carriers in all 
  country...
  

Re: [Asterisk-Users] R2 support

[CW_ASN - Gus]

 Ok, it's old and clunky, but in some countries like Brazil, Argentina and
 China is the only alternative.
Only alternative??? Why is the only alternative? All mayor carriers in
Argentina and Brasil have PRI signalling, at the same price.



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Re: [Asterisk-Users] R2 support

[CW_ASN]

 yes but PRI is not a trunk,

Not in all switches...
You have a Siemens EWSD (I know your company), if you change to V15 you can
treat the PRI like a route (and a lot of things more).
I have Siemens EWSD and Lucent 5ESS, and for 5ESS, the PRI is a route.

I see only one reason to use R2... only when you want to replace old PBX
without change the signalling in CO side.

 R2 can be used as a trunk.
So what? We don't use R2 for trunk purposes since 5 years or more, using PRI
and S7.


Regards,

Gustavo






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Re: [Asterisk-Users] SS7 over Asterisk ?

[CW_ASN - Gus]

See http://www.rad.com/ , TDM-over-IP solutions.


- Original Message -
From: Alexandru Coseru [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, January 19, 2004 6:56 AM
Subject: Re: [Asterisk-Users] SS7 over Asterisk ?


 Maybe , I never tried TDMoE ...
 Where can I found a documentation or at least a sample for doing that ?

 Second , there is a small problem...  Their are not on the same subnet,
but
 this can be fixed(i hope) using tunneling..

 Regards
 Alex


 - Original Message -
 From: Nicolas Bougues [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Monday, January 19, 2004 10:05 AM
 Subject: Re: [Asterisk-Users] SS7 over Asterisk ?


  On Sat, Jan 17, 2004 at 04:34:34PM +0200, Alexandru Coseru wrote:
  
All I'm trying right now is to get raw data from the E1  (from each
timeslot) , transmit it to another asterisk server and push it to the
 other
E1..
  
 
  Doesn't TDMoE do that (provided that you're on the same subnet) ?
 
  --
  Nicolas Bougues
  Axialys Interactive
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Re: [Asterisk-Users] No startup after mpg123 install

[CW_ASN]

run * in console mode and send the 
log.

asterisk -cv


  - Original Message - 
  From: 
  Paul 
  To: [EMAIL PROTECTED] 
  
  Sent: Sunday, January 18, 2004 11:31 
  AM
  Subject: [Asterisk-Users] No startup 
  after mpg123 install
  
  
  
  
  After installing mpg123 * will no 
  longer start up. I get the following error.
  
  ERROR[16384]: File asterisk.c, 
  Line 1349 (main): Unable to connect to remote asterisk
  
  If I remove mpg123, * will run as 
  usual. Any ideas?
  
  ~paul

Re: [Asterisk-Users] Issue - vmail.cgi on Redhat 9 (Apache) ?

[CW_ASN]

Try with:

make webvmail

from source directory.


  - Original Message - 
  From: 
  tony banks 
  To: [EMAIL PROTECTED] 
  
  Sent: Monday, January 12, 2004 1:45 
  PM
  Subject: [Asterisk-Users] Issue - 
  vmail.cgi on Redhat 9 (Apache) ?
  
  HelloI found related question on vmail.cgi in the mailing list but 
  that didn't answer my question. I did copy the vmail.cgi to /var/www/cgi-bin/ 
  but still gets the following error message when I access http://XXX.XX.XX.XXX/cgi-bin/vmail.cgi 
   The server encountered an internal error and was unable to complete 
  your request.   Error message:  Premature 
  end of script headers: vmail.cgi Please adviseRegardsTony 
   

Re: [Asterisk-Users] At last!!! :)

[CW_ASN]

Jess:

Try with:

Dial(SIP/[EMAIL PROTECTED],20,t)

Remove 'r' option from your dial command, maybe 'show application Dial' from
CLI could help you more.

Regards,

Gus


- Original Message -
From: Jess Magnaye
To: [EMAIL PROTECTED]
Sent: Friday, January 09, 2004 7:55 PM
Subject: [Asterisk-Users] At last!!! :)


I can smile now.  I made my * work with my Cisco. Finally.  First problem
was Ethernet on my Linux.  After installing * on a different machine, I got
rid of that icmp udp unreachable error.  My next problem was the call
stays on on Cisco gateway, but the ATA drops it.  I figured out it was my
mistake on dialplan in extensions.conf --- (it took me a day to notice it..
damn!).  my config was: exten=_.,1,Dial(SIP/[EMAIL PROTECTED],tr).  The
reason why my ATA is getting fast busy (or dropping the call immediately)
while Cisco gateway (myprovider) is trying to connect my call, was that I am
missing the seconds parameter.  When I changed this to
Dial(SIP/[EMAIL PROTECTED],20,tr), I was able to connect.

There is one little problem left though.  How come after I diale the number
from ATA, I am getting false ringback.  I meant, local ringback from ATA,
instead of the ringback coming from my Cisco (myprovider).

I appreciate any bright ideas and advise from anybody.

Thank you and have a happy weekend!



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Re: [Asterisk-Users] max queue time; newbie question

[CW_ASN - Gus]

Sure, declare the queue and its timeout, then declare the same extension
with voicemail with n+1 priority.

exten = 2056,1,Answer
exten = 2056,2,Wait,1
exten = 2056,3,Queue(noc|t|||30)
exten = 2056,4,VoiceMail(u2056)

Hope this helps,

Gus

  -= Info about application 'Queue' =-

[Synopsis]:
Queue a call for a call queue

[Description]:
  Queue(queuename[|options[|URL][|announceoverride][|timeout]]):
Queues an incoming call in a particular call queue as defined in
queues.conf.
  This application returns -1 if the originating channel hangs up, or if the
call is bridged and  either of the parties in the bridge terminate the call.
Returns 0 if the queue is full, nonexistant, or has no members.
The option string may contain zero or more of the following characters:
  't' -- allow the called user transfer the calling user
  'T' -- to allow the calling user to transfer the call.
  'd' -- data-quality (modem) call (minimum delay).
  'H' -- allow caller to hang up by hitting *.
  'n' -- no retries on the timeout; will exit this application and go to
the
 next step.
  In addition to transferring the call, a call may be parked and then picked
up by another user.
  The optionnal URL will be sent to the called party if the channel supports
it.
  The timeout will cause the queue to fail out after a specified number of
seconds, checked between each queues.conf 'timeout' and 'retry' cycle.


- Original Message -
From: Ken Alker [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, January 09, 2004 8:07 AM
Subject: [Asterisk-Users] max queue time; newbie question


 I am just studying Asterisk now and have a question.  Is it possible to
 force anyone who enters a queue into voice mail after they have been in
the
 queue for 30 seconds?

 /**
  Ken Alker [EMAIL PROTECTED]ham radio: KA6SDU
  Impulse Internet Services   http://www.impulse.net
  Santa Barbara,  San Luis Obispo,  Ventura, Los Angeles, Orange
  T-3 / T-1 / ADSL / ISDN / 56K / web hosting / wireless / co-lo
 ***/
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Re: [Asterisk-Users] Screen Pop Remote Agents

[CW_ASN - Gus]

snip

 Yes - the Wiki link above about call queues has the info and links that
 you need to look at.
Also, could be great is you install a new patch, to add some great
functionalities to your call center. This path is located:
http://bugs.digium.com/bug_view_page.php?bug_id=214

Regards,

Gus




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Re: [Asterisk-Users] ATA call

[CW_ASN - Gus]

Sometimes Cisco nat changes the port, and * can't contact to ATA. I see that
behavior some days ago, and I can't resolve that. :(

Regards,

Gus

- Original Message -
From: Osvaldo Mundim Junior [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, January 06, 2004 9:15 PM
Subject: Re: [Asterisk-Users] ATA call


 Some times the sip show peers shows me:
 Name/usernameHost Mask Port Status
 porto/porto  (Unspecified)   (D)  255.255.255.255  0UNKNOWN


 and some times shows me:

 Name/usernameHost Mask Port Status
 porto/porto  200.167.103.219 (D)  255.255.255.255  1025 LAGGED
(815
 ms)

 Does the port supposed to be 5060?

 Oz


 - Original Message -
 From: Doug Shubert [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Tuesday, January 06, 2004 9:09 AM
 Subject: Re: [Asterisk-Users] ATA call


  Is your ATA running SIP if so, what version (2.16?)
 
  With SIP, then * extensions.conf and sip.conf files are configured
  you should see the following
 
  asterisk3*CLI sip show peers
  Name/usernameHost Mask Port Status
  3000/300010.0.0.30   (D)  255.255.255.255  5060 OK (15
ms)
  9000/900010.0.0.90   (D)  255.255.255.255  5060 OK (47
ms)
 
  ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960
 
  to test an extension from the CLI
  CLIdial ext. #
  you should hear your ATA ring
 
  Doug
 
  Osvaldo Mundim Junior wrote:
 
   Hey all!
  
   I'm having problems trying to set up an ATA 186 with my Asterisk box.
 When I
   get the phone to place the call, I type the extension and I only get
 busy
   signal after 5 seconds. So I can't call my Asterisk box from my ATA
and
   either call from my Asterisk to my ATA.
  
   Does anybody know what can be happing?
  
   Log is attached..
  
   tks
   regards
   Oz
  
 
   
 Name: ast_log.txt
  ast_log.txtType: Plain Text (text/plain)
 Encoding: quoted-printable
 
  --
  FREE Unlimited Worldwide Voip calling
  set-up an account and start saving today!
  http://www.voippages.com ext. 7000
  http://www.pulver.com/fwd/ ext. 83740
  free IP phone software @
  http://www.xten.com/
  http://iaxclient.sourceforge.net/iaxcomm/
 
 
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Re: [Asterisk-Users] ATA call

[CW_ASN]

Are you using 1605 to do nat?

- Original Message -
From: Eric Wieling [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, January 06, 2004 7:12 PM
Subject: Re: [Asterisk-Users] ATA call


 I have ZERO problems with Cisco's NAT for SIP.

 On Tue, 2004-01-06 at 13:42, CW_ASN - Gus wrote:
  Sometimes Cisco nat changes the port, and * can't contact to ATA. I see
that
  behavior some days ago, and I can't resolve that. :(
 
  Regards,
 
  Gus
 
  - Original Message -
  From: Osvaldo Mundim Junior [EMAIL PROTECTED]
  To: [EMAIL PROTECTED]
  Sent: Tuesday, January 06, 2004 9:15 PM
  Subject: Re: [Asterisk-Users] ATA call
 
 
   Some times the sip show peers shows me:
   Name/usernameHost Mask Port Status
   porto/porto  (Unspecified)   (D)  255.255.255.255  0
UNKNOWN
  
  
   and some times shows me:
  
   Name/usernameHost Mask Port Status
   porto/porto  200.167.103.219 (D)  255.255.255.255  1025 LAGGED
  (815
   ms)
  
   Does the port supposed to be 5060?
  
   Oz
  
  
   - Original Message -
   From: Doug Shubert [EMAIL PROTECTED]
   To: [EMAIL PROTECTED]
   Sent: Tuesday, January 06, 2004 9:09 AM
   Subject: Re: [Asterisk-Users] ATA call
  
  
Is your ATA running SIP if so, what version (2.16?)
   
With SIP, then * extensions.conf and sip.conf files are configured
you should see the following
   
asterisk3*CLI sip show peers
Name/usernameHost Mask Port
Status
3000/300010.0.0.30   (D)  255.255.255.255  5060 OK
(15
  ms)
9000/900010.0.0.90   (D)  255.255.255.255  5060 OK
(47
  ms)
   
ext 3000 is the Cisco ATA 186 and ext 9000 is the Cisco 7960
   
to test an extension from the CLI
CLIdial ext. #
you should hear your ATA ring
   
Doug
   
Osvaldo Mundim Junior wrote:
   
 Hey all!

 I'm having problems trying to set up an ATA 186 with my Asterisk
box.
   When I
 get the phone to place the call, I type the extension and I only
get
   busy
 signal after 5 seconds. So I can't call my Asterisk box from my
ATA
  and
 either call from my Asterisk to my ATA.

 Does anybody know what can be happing?

 Log is attached..

 tks
 regards
 Oz

   
 
  
   Name: ast_log.txt
ast_log.txtType: Plain Text (text/plain)
   Encoding: quoted-printable
   
--
FREE Unlimited Worldwide Voip calling
set-up an account and start saving today!
http://www.voippages.com ext. 7000
http://www.pulver.com/fwd/ ext. 83740
free IP phone software @
http://www.xten.com/
http://iaxclient.sourceforge.net/iaxcomm/
   
   
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 --
 Go to http://www.digium.com/index.php?menu=documentation and look at
 the Unofficial Links section.  This section has links to a wide
 variety of 3rd party Asterisk related pages.  My page is the
 Asterisk Resource Pages.

 BTEL Consulting 504-899-1387 or 850-484-4545 or 877-677-9643

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Re: [Asterisk-Users] Hpw to enable Voicemail Indicator on IP/Analog Phone ?

[CW_ASN]

snip
 this is called Message Waiting Indicator (MWI) in asterisk.  I
 haven't set it up myself,
 but from what I've seen there are a few parts:

 1) setting a mailbox=1234 etc. in the extension definition in the channel
 file
 2) setting up the phone

 Have a look around the wiki
 http://www.voip-info.org/tiki-index.php?page=SIP%20mwi

 It says CISCO works
http://www.voip-info.org/wiki-Asterisk+phone+cisco+79xx
 and ADSI, and grandstream,
 I assume SNOM can, it's just not documented on the wiki yet, maybe you can
 contribute that :-)

 Cheers,
 Woody

That's true. In SIP, when user registers, Asterisk indicates how many
messages you have... In ATA18x equipments, it's indicates by stutter tone.

Regards,

Gus






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Re: [Asterisk-Users] SIP/grandstream not registering

[CW_ASN]

And why you have two different entries for the same object?
Posting two times the same questions with other data will not help to
resolve the issue more quickly...

- Original Message -
From: Glenn Dalgliesh [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED]
Sent: Saturday, January 03, 2004 4:00 AM
Subject: Re: [Asterisk-Users] SIP/grandstream not registering


 It looks like you have you * on public IP and your phones on private, most
 likely behind NAT if so in your sip.conf under each [grandstreamX] you
most
 likely need:  nat=yes


 - Original Message -
 From: Chandra [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Saturday, January 03, 2004 1:44 AM
 Subject: [Asterisk-Users] SIP/grandstream not registering


  hi,
 
  i can't seem to register my grandstream SIP to * server...
 
  i have my grandstream IP as 192.168.0.11 want to register to * at
  202.51.xx.xxx.
 
  sip show peers says that my grand stream has unspecified IP but when i
try
  to dial a number it gets this error...
  WARNING[5126]: File chan_sip.c, Line 464 (retrans_pkt): Maximum retries
  exceeded on call [EMAIL PROTECTED] for
 seqno
  40939 (Response)
 
 
  my sip.conf is...
  [grandstream2]
  type=peer
  host=dynamic
  secret=grandstream2
  reinvite=no
  canreinvite=no
  qualify=60
 
 
  [grandstream2]
  type=user
  host=dynamic
  secret=grandstream2
  context=outgoing
  reinvite=no
  canreinvite=no
  qualify=60
 
  help
 
 
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Re: [Asterisk-Users] Call recording/SIP not loggin IN

[CW_ASN]

- Original Message -
From: Chandra [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 03, 2004 1:34 AM
Subject: Re: [Asterisk-Users] Call recording/SIP not loggin IN


 My sip.conf
 [general]
 port = 5060 ; Port to bind to
 bindaddr = 0.0.0.0
 disallow=all; Disallow all codecs
 allow=ulaw  ; Allow codecs in order of preference

 dtmfmode=rfc2833

 [xlite1]
 type=user
 host=dynamic
 secret=xlite1
 context=outgoing
 reinvite=no
 canreinvite=no
 qualify=60

 [xlite1]
 type=peer
 host=dynamic
 secret=xlite1
 reinvite=no
 canreinvite=no
 qualify=60

 In xlite i have User=xlite1, Pwd=xlite1 and SIP Proxy=IP of my * box, Out
 bound Proxy= IP of my * box

 netstat -na gives

 [EMAIL PROTECTED] root]# netstat -na
 Active Internet connections (servers and established)
 Proto Recv-Q Send-Q Local Address   Foreign Address State
 tcp0  0 0.0.0.0:32768   0.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:22305   0.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:22273   0.0.0.0:*   LISTEN
 tcp0  0 127.0.0.1:32769 0.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:33060.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:111 0.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:20000.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:56800.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:80  0.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:22321   0.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:22289   0.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:21  0.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:22  0.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:23  0.0.0.0:*   LISTEN
 tcp0  0 0.0.0.0:443 0.0.0.0:*   LISTEN
 tcp0128 202.51.xx.xx1:22202.51.xx.xx0:3148
 ESTABLISHED
 udp0  0 0.0.0.0:32769   0.0.0.0:*
 udp0  0 0.0.0.0:50360.0.0.0:*
 udp0  0 0.0.0.0:50600.0.0.0:*
 udp0  0 0.0.0.0:45690.0.0.0:*
 udp0  0 0.0.0.0:111 0.0.0.0:*
 udp0  0 0.0.0.0:11770   0.0.0.0:*
 udp0  0 0.0.0.0:11771   0.0.0.0:*
 udp0  0 0.0.0.0:24270.0.0.0:*
 Active UNIX domain sockets (servers and established)
 Proto RefCnt Flags   Type   State I-Node Path
 unix  2  [ ACC ] STREAM LISTENING 1504   /dev/gpmctl
 unix  2  [ ACC ] STREAM LISTENING 1775
 /tmp/.font-unix/fs7100
 unix  2  [ ACC ] STREAM LISTENING 1520
 /var/lib/mysql/mysql.sock
 unix  2  [ ACC ] STREAM LISTENING 1885
 /var/run/asterisk.ctl
 unix  2  [ ACC ] STREAM LISTENING 1621
 /tmp/.iroha_unix/IROHA
 unix  2  [ ACC ] STREAM LISTENING 1593   /tmp/cd_sockV4
 unix  2  [ ACC ] STREAM LISTENING 1671   /tmp/kd_sockV4
 unix  2  [ ACC ] STREAM LISTENING 1699   /tmp/td_sockV4
 unix  2  [ ACC ] STREAM LISTENING 1565   /tmp/jd_sockV4
 unix  7  [ ] DGRAM1094   /dev/log
 unix  3  [ ] STREAM CONNECTED 1889
 /var/lib/mysql/mysql.sock
 unix  3  [ ] STREAM CONNECTED 1888
 unix  2  [ ] DGRAM1778
 unix  2  [ ] DGRAM1645
 unix  2  [ ] DGRAM1406
 unix  2  [ ] DGRAM1160
 unix  2  [ ] DGRAM1110
 [EMAIL PROTECTED] root]#


 my grandstream is also not registering to *.


You have two entries for [xlite1].
In order to test, first remove 'qualify' and 'reinvite' from the sip.conf,
reload and try again.
If you don't use NAT, then you should delete OutBoundProxy from xlite
config., and set 'Use OutboundProxy' as 'Never'.
Make sure that xlite is setted as Send internal IP Always.

Assuming that you have only one IP address (and a loopback) in your box,
'netstat' looks good.

Next steps could be dump the traces in xlites, and * box, to see whats wrong
more deeply.


Hope this helps, please advice.

Gus




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Re: [Asterisk-Users] New to asterisk? RUN... don't walk.

[CW_ASN]

If you are a person who likes all things easy, and if you don't need to know
nothing to be better professional, well, run now, and let us continue our
journey. Who cares? People likes you don't help to our community.

Regards,

Gus

- Original Message -
From: Me [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 31, 2003 5:37 PM
Subject: [Asterisk-Users] New to asterisk? RUN... don't walk.


 As a newcomer to Asterisk, you will not be welcomed
 with open arms.  First, you will find almost no
 documentation on it's features.  Second, if you try to
 ask questions, you will be flamed and pointed to
 worthless how-tos and 'the wiki'.  These worthless
 documents can only be useful for explaining how things
 work to those already in-the-know.  Lastly, Asterisk
 is so bug ridden, expect frequent segmentation faults.
  With a community so 'anti-n00b', don't expect your
 problems to be fixed anytime soon.

 RUN!!! Don't walk... away from Aterisk.

 __
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 Find out what made the Top Yahoo! Searches of 2003
 http://search.yahoo.com/top2003
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Re: [Asterisk-Users] New to asterisk? RUN... don't walk.

[CW_ASN]

 Dear newbies,

 As a newcomer to woodworking, you will not be welcomed with open arms.
 First, you will find no documentation on how to make your completely
custom
 ceiling-height cabinets perfectly the first time that your wife will
 appreciate. Second, if you ask any woodworker for assistance, you will be
 treated like a fool and your new cabinets will be set aflame and you will
be
 instructed to experiment with your tool and learn your craft. This
worthless
 waste of time will only develop you into a competent woodworker able to
make
 anything you wish. You should go to the furniture store or ask an already
 competent person to take care of your cabinetry for you as you have
neither
 the desire or intelligence. Lastly, your raw material is so bug-ridden,
all
 your handiwork will prove fruitless. We should all leave it up to the
 experts. With a carpentry community so anti-n00b, don't expect your
 handbuilt cabinets to be fixed for free by other people with their own
 problems who have graciously given their time and knowledge to the rest of
 us. You might actually be expected to fix it yourself.

Certainly great! You make me laugh so much...



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Re: [Asterisk-Users] call pickup via *8 from ata186 (SIP)

[CW_ASN]

It works for me with sip 2.15, 2.16.x and 3 versions.

- Original Message - 
From: Anton Yurchenko [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, December 29, 2003 6:42 AM
Subject: [Asterisk-Users] call pickup via *8 from ata186 (SIP)


 Hello,
 
 Does call pickup works with ATA-186 SIP? at the same pbx it works with 
 MGCP but bit ata-186 with SIP it doesnt work, just nothing happens. 
 Anyone have it working? Also it seems that when typing reload on the 
 console, the asterisk doesnt reread the mgcp.conf.
 
 please answer
 
 Thanks
 
 -- 
 
 Anton Yurchenko[EMAIL PROTECTED]
 Digital Generation
 
 
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Re: [Asterisk-Users] Agent setup

[CW_ASN]

Shad:

Using the AddQueueMember. Launching this command 3 
times in different queues, logs one phone to that 3 queues...

*CLI show application AddQueueMember

 -= Info about application 'AddQueueMember' =-

[Synopsis]:Dynamically adds queue members

[Description]: 
AddQueueMember(queuename[|interface]):Dynamically adds interface to an 
existing queueReturns -1 if there is an error.Example: 
AddQueueMember(techsupport|SIP/3000)

*CLI
Or, you must declare only oneextension for 
all agents, i.e:

;LogOn and LogOff in queue: colaexten = 
icola,1,EAGI(opinc.php,cola)exten = icola,2,Hangupexten = 
dcola,1,EAGI(opdel.php,cola)exten = dcola,2,Hangup

The AGIs contains:

--- opinc.php:

#!/usr/bin/php 
-q?phpob_implicit_flush(true);set_time_limit(0);

function sm($texto) {

echo("VERBOSE \"".$texto."\"\n");

}

function se($comando) {

echo("exec $comando\n");

}

//MAIN PROCEDURE

$err = fopen("php://stderr","w");$in = 
fopen("php://stdin","r");

while (!feof($in)) 
{ $temp = 
str_replace("\n","",fgets($in,4096)); 
$s = split(":",$temp); 
$agi[str_replace("agi_","",$s[0])] = trim($s[1]);

 if 
(($temp == "") || ($temp == "\n")) 
{ 
break; }}

$cid=trim($agi["callerid"]);$ext=trim($agi["extension"]);$tec=trim($agi["type"]);

$rt1=strpos($cid,"");$rt2=strpos($cid,"");if 
(strpos($cid,"")0) { 
$cid=trim(substr($cid,($rt1+1),(($rt2-$rt1)-1)));} else 
{ 
$cid=trim($cid);}

$cid=stripslashes($cid);

$cola=$argv[1];

se("AddQueueMember $cola 
$tec/$cid");se("Playback agent-loginok");sm("Agent '$tec/$cid' was 
included in queue '$cola'");

?

--- opdel.php:

#!/usr/bin/php 
-q?phpob_implicit_flush(true);set_time_limit(0);

function sm($texto) {

echo("VERBOSE \"".$texto."\"\n");

}

function se($comando) {

echo("exec $comando\n");

}

//MAIN PROCEDURE

$err = fopen("php://stderr","w");$in = 
fopen("php://stdin","r");

while (!feof($in)) 
{ $temp = 
str_replace("\n","",fgets($in,4096)); 
$s = split(":",$temp); 
$agi[str_replace("agi_","",$s[0])] = trim($s[1]);

 if 
(($temp == "") || ($temp == "\n")) 
{ 
break; }}

$cid=trim($agi["callerid"]);$ext=trim($agi["extension"]);$tec=trim($agi["type"]);

$rt1=strpos($cid,"");$rt2=strpos($cid,"");if 
(strpos($cid,"")0) { 
$cid=trim(substr($cid,($rt1+1),(($rt2-$rt1)-1)));} else 
{ 
$cid=trim($cid);}

$cid=stripslashes($cid);

$cola=$argv[1];

se("RemoveQueueMember $cola 
$tec/$cid");se("Playback agent-loggedoff");sm("Agent '$tec/$cid' was 
removed from queue '$cola'");

?
I know, the code is dirty... but it works for 
me.

Hope this helps,

Regards,

Gus



  - Original Message - 
  From: 
  Shad Mortazavi 
  To: '[EMAIL PROTECTED]' 
  
  Sent: Monday, December 29, 2003 4:50 
  PM
  Subject: [Asterisk-Users] Agent 
  setup
  
  
  Dear 
  Group,
  
  I have been successful in setting 
  up the Agents, queues and getting agents to log 
  in.
  
  Is there a way that I could 
  configure the system so that the agent is called back. i.e. the agent logs 
  into the system, a call is destined for them and their phone 
  rings.
  
  If some one has this setup I would 
  be very interested in hearing from them.
  
  Warm Regards and 
  Thanks
  
  ---
  Shad 
  Mortazavi
  US Technical 
  Manager
  Nexus 
  Management
  
  

Re: [Asterisk-Users] Agent setup

[CW_ASN]

 Easier but poorly documented solution. AgentCallbackLogin()

 AgentCallbackLogin delivers callo for a logged in agent to an extension.
 - they continue to get calls until they log out (by logging in to a null
 extension (pressing # when prompted for extension)

But AgentCallbackLogin remains the line active all the time. Is the same
behavior than AgentLogin, but * hangs up and call back to the original user.

*CLI show application AgentCallbackLogin

  -= Info about application 'AgentCallbackLogin' =-

[Synopsis]:
Call agent callback login

[Description]:
  AgentCallbackLogin([AgentNo][|[EMAIL PROTECTED]):
Asks the agent to login to the system with callback.  Always returns -1.
The agent's callback extension is called (optionally with the specified
context.

*CLI



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Re: [Asterisk-Users] prepaid app

[CW_ASN]

Oh, Jeremy... here we goes again...

First:
In that time, Bartosz distribute this app to test purposes only... please
read the list messages... (DEC/01)

Second:
This app makes a rating better than other commercial products (and I know
enough on this, and I see a lot of platforms)... Touching some lines of
code, this app runs like a heaven.

Third:
I know unix-odbc exists (thanks to Brian), but app_prepaid is based on
postgres... Why? I don't know... I'm not a developer of app_prepaid. I'm not
see the devil inside postgres...


Regards,

Gus


- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 26, 2003 6:10 PM
Subject: Re: [Asterisk-Users] prepaid app


 CW_ASN wrote:

 Send an email to Bartosz, he has app_prepaid. You will need to work a lot
 with C (i'm learning) and pgsql, but is very nice app.
 
 
 


 First off he cannot distribute his C API based app without 1) releasing
 it GPL or 2) paying Digium for a non-gpl licnese.

 Secondly, I seriously hope the application itself doesn't actually to
 the call rating.

 Thirdly, why dictate what DB can get used?  unix-odbc has gotten a whole
 lot better in the last year. Plus the API is lot more forgiving and
 won't core your box, if the developer hasn't tested for every possible
 situation.


 Jeremy McNamara


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Re: [Asterisk-Users] prepaid app

[CW_ASN]

 Doesn't matter.  If he uses the C API he he bound by the GPL or he has
 to pay digium's fees for non-gpl.

Who in the hell said that is not GPL? I'm not sure about the licence of this
app, but in the .c code shows a nice GPL...
Maybe this 2 lines makes your life easier...

 * This program is free software, distributed under the terms of
 * the GNU General Public License




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Re: [Asterisk-Users] time to build an open phone?

[CW_ASN]

How about to build an ip phone with this IC?

http://focus.ti.com/docs/apps/catalog/general/applications.jhtml?templateId=
969path=templatedata/cm/general/data/bband_ipphone_tnetv1001


- Original Message -
From: Bob Knight [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, December 24, 2003 4:30 PM
Subject: [Asterisk-Users] time to build an open phone?


 Open software seems to work.
 Why don't we try it with hardware.

 1. pick an embedded processor.
 It should have a nice linux gui support (like x jtag debugger).

 2. pick a linux based cad system we all have easy access to and place
 schematics under cvs.

 3. pick some type of gpio or serial interface for keyboard/display.

 4. pick some basic functionality.

 5. code it up. A stripped down *.

 Let everyone do their own thing with the expensive part.
 Tooling/packaging.

 We could let Digium be the distributor, so they are not left out of the
 loop.
 A board set would be offered with NO support.
 If Digium wants no part of it, we just build them on our own for our own
use
 or sell them on ebay.

 What we would provide is schematics and source code.
 Everyone can take this to their favorite fab house and crank em out.

 --
 Bob Knight
 [-w] the work option
 [EMAIL PROTECTED]
 925-449-9163


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Re: [Asterisk-Users] OT: SIP vs. Skinny protocol

[CW_ASN]

 Skinny phone functionality is 'richer' than SIP phone functionality.
First
 off, on a skinny phone, in hands free mode, you can start dialling and the
 phone will automatically go off hook.  Sip requires you to manually hit
the
 speaker button, hit new call, or pickup the phone before dialling.  (One
 extra confusing key stroke I have a hard time getting over).

This is not a sip issue, it's a phone funcionality...


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Re: [Asterisk-Users] Asterisk + CRM

[CW_ASN - Gus]

Which events do you refer?

Regards,

Gus

- Original Message - 
From: Jonathan Tew [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 23, 2003 12:25 PM
Subject: Re: [Asterisk-Users] Asterisk + CRM


 We're starting to integrate * with our customer service software.  
 Basically we're pulling off events from the management interface.  We're 
 also making some small patches to the code to deliver more events about 
 the channel variables, etc. 
 
 Anton Yurchenko wrote:
 
  Hello,
 
  Anyone aware of any CRM products projects that intagrete with *? Or 
  that integrate with any telephony products? Is there some open API for 
  such integration, or are they all proprietory?
 
  Thanks
 
 
 
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Re: [Asterisk-Users] RE: voicemail file permissions

[CW_ASN - Gus]

Guys, I'm using RH9 with vmail.cgi without any modifications... I'm just do
a 'make webvmail' after 'make install'... I don't have any troubles...

Regards,

Gus

- Original Message -
From: Carlton J. O'Riley [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 04, 2003 4:23 PM
Subject: [Asterisk-Users] RE: voicemail file permissions


Here is a script I use in a cron job that runs every 5 minutes to make it so
that my webserver (which runs as the apache group) can access the voicemails
through the web.  Seems to fix my problems.  Although if I get the email
there is a voicemail it might be 5 minutes before I can get to it via the
web, but you could increase the frequency at which this job runs.  Five
minutes has been fine for me.  It'd be nice to be able to set the owner and
group and permissions for voicemail files in the configuration file for
voicemail.  If I had time I'd probably do it myself.

Carlton

#!/bin/sh
/bin/chgrp -R apache /var/spool/asterisk/vm/*
/bin/chmod -R g+rw /var/spool/asterisk/vm/*

  hi, i realised that when voicemails are recorded it is set to 700 file =
permission and which leads to a serrious problem when
accessing the = voicemail thru the web using vmail,cgi

  how can i automatically set the file permission to 755 or 777 so that = i
can make it readeable from the web? which file in * helps
 to record = the voicemail and create that voicemail in a certain dir?? if
any onw = knows, i can perhaps find that line and change as  nesseciate.

  anyone tried vmail.cgi could help.
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Re: [Asterisk-Users] Meetme Recording

[CW_ASN - Gus]

Try something like this:

exten = 2060,1,Answer
exten = 2060,2,Wait,1
exten = 2060,3,Monitor,wav|algo
exten = 2060,4,Meetme,1|ps

Regards,

Gus

- Original Message -
From: Girish Gopinath [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 8:58 AM
Subject: [Asterisk-Users] Meetme Recording


 Hi,

 Can anybody explain me in configuring Asterisk to record a conference?

 Regards...

 Girish

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Re: [Asterisk-Users] Meetme Recording

[CW_ASN - Gus]

algo is a file where app write a wav data. In spanish, algo means
something... :)

Gus

  -= Info about application 'Monitor' =-

[Synopsis]:
Monitor a channel

[Description]:
Monitor
Used to start monitoring a channel. The channel's input and output
voice packets are logged to files until the channel hangs up or
monitoring is stopped by the StopMonitor application.
The option string may contain the following arguments:
[file_format|[fname_base]]
file_format -- optional, if not set, defaults to wav
fname_base -- if set, changes the filename used to the one
specified.


- Original Message -
From: Dave Packham [EMAIL PROTECTED]
To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 1:59 PM
Subject: Re: [Asterisk-Users] Meetme Recording


what do the options algo do in the monitor app?  I dont see that in the
show application monitor?   is this a patch?

Dave

 [EMAIL PROTECTED] 12/2/2003 6:56:18 AM 
Try something like this:

exten = 2060,1,Answer
exten = 2060,2,Wait,1
exten = 2060,3,Monitor,wav|algo
exten = 2060,4,Meetme,1|ps

Regards,

Gus

- Original Message -
From: Girish Gopinath [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Tuesday, December 02, 2003 8:58 AM
Subject: [Asterisk-Users] Meetme Recording


 Hi,

 Can anybody explain me in configuring Asterisk to record a conference?

 Regards...

 Girish

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Re: [Asterisk-Users] Re: Unable to use voicemail(Please suggestion)

[CW_ASN - Gus]

Try with another codec different than G.723. Use GSM o G.711 for this.
You could disable G.723 in your sip.conf

disallow=all
allow=gsm
allow=alaw
allow=ulaw

Hope this helps,

Gus

- Original Message -
From: Hachy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 12, 2003 12:32 AM
Subject: [Asterisk-Users] Re: Unable to use voicemail(Please suggestion)


 Hello all

 I got call log from Asterisk.
 I call to ext1001 from ext1002.
 But could not leave a message in the voice mail.

 Please help me.

 -- Executing Dial(SIP/1002-8217, SIP/1001|20) in new stack
 -- Called 1001
 -- SIP/1001-25ce is ringing
 -- Nobody picked up in 2 ms
   == Spawn extension (sip, 1001, 2) exited non-zero on 'SIP/1002-8217



 
 Hello all.
 
 Now I aleady installed the Asterisk.
 I could make communication between 2 XLite client through Asterisk.
 
 I tryed to test the voicemail function as follow.
  1, I make a call to 1001 from 1002
  2, Start ringing
  3, Wait untill time out for ringing
 
 If no problem, 1001 go to voicemail and unavailable message will
 be played.
 But 1001 receive a 403 forbidden massage and connection go down.
 And Icould not leave a messages.
 Please teach me how to resolve this problem.
 
 Here is configuration of Asterisk and Xlite.
 #sip.conf in Asterisk
 [general]
 port=5060
 bindaddr=0.0.0.0
 nortifymimetype=text/plain
 allow=all
 [1001]
 type=friend
 username=1001
 secret=1001
 host=dynamic
 defaultip=192.168.0.1
 mailbox=1001
 context=sip
 canreinvite=no
 [1002]
 type=friend
 username=1002
 secret=1002
 host=dynamic
 defaultip=192.168.0.1
 mailbox=1002
 context=sip
 canreinvite=no
 
 #extensions.conf in Asterisk
 [general]
 static=yes
 writeprotect=no
 [glovals]
 CONSOLE=Console/dsp
 [sip]
 exten = 1001,1,Dial(SIP/1001,20)
 exten = 1001,2,Voicemail(u1001)
 exten = 1001,102,Voicemail(b1001)
 exten = 1001,103,Hungup
 exten = 1002,1,Dial(SIP/1001,20)
 exten = 1002,2,Voicemail(u1002)
 exten = 1002,102,Voicemail(b1002)
 exten = 1002,103,Hungup
 
 #voicemail.conf in Asterisk
 [local]
 1001 = 1001,1001,mail address
 1002 = 1002,1002,mail address
 
 #Create mailbox by addmailbox already.
 
 #Client configuration
 User Name1001   1002
 Authorization User   same as username
 PAssword 1001   1002
 Domain/Realm 192.168.0.120
 SIP Proxy192.168.0.120
 
 Here is call flow on this test.
 
 (c)2003 Xten Networks Inc. All rights reserved.
 Private build: 1008
 SIP: 192.168.0.125:5061
 RTP: 192.168.0.125:8000
 NAT: 210.253.186.126
 PXY#0: 192.168.0.120:5060
 
 RECEIVE  192.168.0.120:5060
 NOTIFY sip:[EMAIL PROTECTED]:5061 SIP/2.0
 Via: SIP/2.0/UDP 192.168.0.120:5060;branch=z9hG4bK375605f3
 From: asterisk sip:[EMAIL PROTECTED];tag=as633f7afa
 To: sip:[EMAIL PROTECTED]:5061
 Contact: sip:[EMAIL PROTECTED]
 Call-ID: [EMAIL PROTECTED]
 CSeq: 102 NOTIFY
 User-Agent: Asterisk PBX
 Event: message-summary
 Content-Type: text/plain
 Content-Length: 36
 Messages-Waiting: no
 Voicemail: 0/0
 
 SEND  192.168.0.120:5060
 INVITE sip:[EMAIL PROTECTED] SIP/2.0
 Via: SIP/2.0/UDP 192.168.0.125:5061
 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961
 To: sip:[EMAIL PROTECTED]
 Contact: sip:[EMAIL PROTECTED]:5061
 Call-ID: [EMAIL PROTECTED]
 CSeq: 26502 INVITE
 Content-Type: application/sdp
 Content-Length: 301
 
 v=0
 o=1002 22002568 22002568 IN IP4 192.168.0.125
 s=X-Lite
 c=IN IP4 192.168.0.125
 t=0 0
 m=audio 8000 RTP/AVP 4 0 8 3 101
 a=rtpmap:4 G723/8000
 a=rtpmap:0 pcmu/8000
 a=rtpmap:8 pcma/8000
 a=rtpmap:3 gsm/8000
 a=rtpmap:101 telephone-event/8000
 a=fmtp:101 0-15
 a=rtpmap:126 x-pro-encrypted/8000
 
 RECEIVE  192.168.0.120:5060
 SIP/2.0 407 Proxy Authentication Required
 Via: SIP/2.0/UDP 192.168.0.125:5061
 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961
 To: sip:[EMAIL PROTECTED];tag=as08d3281f
 Call-ID: [EMAIL PROTECTED]
 CSeq: 26502 INVITE
 User-Agent: Asterisk PBX
 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
 Contact:
 Proxy-Authenticate: Digest realm=asterisk, nonce=05d14468
 Content-Length: 0
 
 
 SEND  192.168.0.120:5060
 ACK sip:[EMAIL PROTECTED] SIP/2.0
 Via: SIP/2.0/UDP 192.168.0.125:5061
 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961
 To: sip:[EMAIL PROTECTED];tag=as08d3281f
 Contact: sip:[EMAIL PROTECTED]:5061
 Call-ID: [EMAIL PROTECTED]
 CSeq: 26502 ACK
 Max-Forwards: 70
 Content-Length: 0
 
 
 SEND  192.168.0.120:5060
 INVITE sip:[EMAIL PROTECTED] SIP/2.0
 Via: SIP/2.0/UDP 192.168.0.125:5061
 From: 1002 sip:[EMAIL PROTECTED]:5061;tag=337011961
 To: sip:[EMAIL PROTECTED]
 Contact: sip:[EMAIL PROTECTED]:5061
 Call-ID: [EMAIL PROTECTED]
 CSeq: 26503 INVITE
 Proxy-Authorization: Digest username=1002,realm=asterisk,nonce=
 05d14468,response=8fb4b56e7dae5665a8ea56a34027be5f,uri=sip:[EMAIL PROTECTED]
 168.0.120
 Content-Type: application/sdp
 Content-Length: 301
 
 v=0
 o=1002 22002778 22002778 IN IP4 192.168.0.125
 s=X-Lite
 c=IN IP4 192.168.0.125
 t=0 0
 m=audio 8000 RTP/AVP 4 0 8 3 101
 

Re: [Asterisk-Users] Media Negotiation Failed

[CW_ASN - Gus]

Title: Mensaje



Fijate en los 'voice codecs' de los 
dial-peers.

  - Original Message - 
  From: 
  Sebastian Nocetti 
  To: [EMAIL PROTECTED] 
  
  Sent: Wednesday, November 12, 2003 12:41 
  PM
  Subject: [Asterisk-Users] Media 
  Negotiation Failed
  
  Hi, I have this 
  scenario
  
  Cisco 5300 (public 
  ip. 200.47.xx.xx) --- Asterisk (public ip: 64.76.xx.xx) -- 
  Cisco 3600 (public ip: 64.76.xx.xx , same network than * )
  
  When a calls comes 
  in Cisco 5300, this send this calls with SIP to *, asterisk plays a welcome 
  message and resend call to Cisco 3600 that have 4 analog lines connected... 
  but after cisco play welcome message and whensend SIP to 3600, I have 
  this error:
  
  v=0o=root 
  20045 20045 IN IP4 64.76.xx.xx - asterisk ip addresss=sessionc=IN 
  IP4 64.76.xx.xx - asterisk ip address.t=0 0m=audio 15372 RTP/AVP 0 
  101a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 
  0-16(no NAT) to 64.76.xx.xx:5060 - 3600 ip addressSip read: 
  LISIP/2.0 400 Bad Request - 'Media Negotiation Failed'Via: 
  SIP/2.0/UDP 64.76.xx.xx:5060;branch=z9hG4bK31ba01da - asterisk ip 
  addressFrom: "1143724956" 
  sip:[EMAIL PROTECTED];tag=as33c45436 - * ip addressTo: 
  sip:[EMAIL PROTECTED] -3600 ip addressCall-ID: [EMAIL PROTECTED]Warning: 
  304 64.76.xx.xx:0 "Media Type(s) Unavailable" - 3600 ip addressCSeq: 
  102 INVITE
  
  then I have too 
  another GW 5300, with same IOS and same config.. and with it, all work 
  OK!!!... I don't understand what is the problem!!...
  
  
  
  IT WORKS 
  OK!!!..
  
  Cisco 5300 (public 
  ip. 64.76.xx.xx) --- Asterisk (public ip: 64.76.xx.xx) -- 
  Cisco 3600 (public ip: 64.76.xx.xx , same network than * )
  
  
  Some 
  clue?

Re: [Asterisk-Users] menu prompts and voice mail greetings.

[CW_ASN]

Did you record the messages as gsm format?

- Original Message - 
From: Larry D. Black [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, November 12, 2003 6:33 PM
Subject: [Asterisk-Users] menu prompts and voice mail greetings.


 What program do you use to record menu prompts and voice mail greetings
 we tried windows recorder and it kept telling us bad file format.
 
 Thanks.
 
 
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Re: [Asterisk-Users] Manager Server

[CW_ASN - Gus]

Yes, is posible.

  - Original Message - 
  From: 
  marin 
  blu 
  To: [EMAIL PROTECTED] 
  
  Sent: Thursday, November 06, 2003 3:22 
  AM
  Subject: [Asterisk-Users] Manager 
  Server
  
  Hi,
  
  Is it possible to control * fromthe TCP Manager Server in order to 
  support CRM systems ?
  
  MarinBlu
  
  
  
  Do you Yahoo!?Protect 
  your identity with Yahoo! Mail AddressGuard

Re: [Asterisk-Users] Voicemail2 vs Voicemail

[CW_ASN - Gus]

Just replace Voicemail by VoiceMail2 and that's all.
Note that new voicemail.conf is a bit different than old voicemail.conf.

Regards,

Gus

- Original Message - 
From: Rich Adamson [EMAIL PROTECTED]
To: Asterisk-a-users-list [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 4:44 AM
Subject: [Asterisk-Users] Voicemail2 vs Voicemail


 
  Wouldn't that break everybody's dialplans where they would have to
  replace all occurrences of Voicemail2 with Voicemail and all
  occurrences of Voicemailmain2 with Voicemailmain?
 
  No, we would register with both names.
 
 Is it necessary (with reasonably current cvs) to make any changes in the
 *.conf files to use Voicemail2, or is that happening automatically?
 
 
 
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Re: [Asterisk-Users] How to control dialout in extensions file

[CW_ASN - Gus]

You could use DISA app.

exten = 2101,1,DISA,/opt/pass.txt|default

Where:
/opt/pass.txt is a plain text file with password list.
default is a destination context.

Anyway, please do 'show application disa' from CLI.

Hope this helps,

Gus

- Original Message - 
From: Jacky Chen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, November 06, 2003 8:18 AM
Subject: [Asterisk-Users] How to control dialout in extensions file


 Hi, all
 
 I have builded a pbx server for pstn, sip  h.323 users
 but i can't find any example extensions.conf for access 
 control when users which call longdistance with pstn,
 
 If anyone have good example, please sharing your experience
 Thanks very much
 
 
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Re: [Asterisk-Users] PHP Manager examples

[CW_ASN]

Here is my example. I'm using a lot of times a day.

?php

$socket = fsockopen(192.168.0.53,5038, $errno, $errstr, $timeout);
fputs($socket, Action: Login\r\n);
fputs($socket, UserName: admin\r\n);
fputs($socket, Secret: blabla\r\n\r\n);

fputs($socket, Action: Command\r\n);
fputs($socket, Command: reload\r\n\r\n);
$wrets=fgets($socket,128);

?

Regards,

Gus

- Original Message -
From: Kevin Bockman [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, November 02, 2003 6:42 PM
Subject: [Asterisk-Users] PHP Manager examples


 Anyone have any example scripts in PHP that connect to the manager?  I'm
not really a much of a programmer so I could use boost.  Once I can figure
out how to get it to login properly, I'll be ok from there.

 Thanks,

 Kevin

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Re: [Asterisk-Users] SS7 signaling/Softswitch

[CW_ASN - Gus]

Brad:

Lucent 5ESS can treat a PRA (PRI) like a CCS7 or R2 route. This feature
allow to route any number, including numbers outside the DID range. I know
some class 5 switches (Lucent, Siemens, Ericsson, Nec, Alcatel and Italtel),
and only Lucent 5ESS and Siemens EWSD provides this feature (at least in
world-wide market, I mean, non US market). If you like to connect a VoIP
gateway, in the 5ESS you can route any number, like local, national or
international. I have * connected to our 5E switch, and it works good.

PRI was defined as the ISDN termination for customer side in 'high-speed'
Nx64 or Nx56; supports all advanced features provided by ISDN and in general
terms, provides full mapping for ISUP layer 4 data.
At least for EuroISDN, for IN and AIN features, PRI only provides voice
connection with SSP switch. PRI doesn't have a special messages replacing or
adding features to INAP, because it haven't a message structure neither
querys nor commands (you can't specify triggers in PRI signalling), only for
ISDN termination related to call establishment.


Regards,

Gus


- Original Message -
From: Brad Waite [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 5:21 PM
Subject: Re: [Asterisk-Users] SS7 signaling/Softswitch


 CW_ASN - Gus wrote:

  Anyway, in certanly implemetations you don't need CCS7 to connect to CO.
You
  always can connect with PRI... same speed and same functionalities to
user
  side. In fact, CCS7 is the support for ISDN-PRI avanced features. If you
  could connect with Lucent 5ESS you can have a PRI treated as route...

 Gus,

 I'm not following you here when you say, ...you can have a PRI treated as
 route...  Can you clarify?

 I'm trying to determine what AIN features may be available on a PRI D
channel.
 I know the D channel is a near extension to SS7, but I don't know what
subset of
   queries/commands are available between the two.

 Brad Waite
 W Cubed

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Re: [Asterisk-Users] Music Onhold Configuration

[CW_ASN - Gus]

I didn't know it... excellent!

- Original Message - 
From: Thorsten Lockert [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 6:36 PM
Subject: RE: [Asterisk-Users] Music Onhold Configuration


  MPG123 is not included in Asterisk...
  Download the package:
  
  http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/precompiled/
  
  Install using:
  
  rpm -ivh mpg123-0.59q-1.i386.rpm
  
  Copy the file mpg123 from /usr/local/bin to /usr/bin
 
 You no longer need to copy it from /usr/local/bin to /usr/bin -- Asterisk
 will look for it in either place.
 
 Thorsten 
 
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Re: [Asterisk-Users] SS7 signaling/Softswitch

[CW_ASN - Gus]

 Close.  Normally, at least in Qwest-land, third-party VM provider systems
dial
 into the switch and give it a DN and a MWI on-or-off command.  If the DN
is
 serviced by that switch, it turns the message waiting indicator (stutter
 dialtone, MW light or both) on or off.  If the number is on a remote
switch, the
 information gets sent over the SS7 network to the other Qwest switch.  I
haven't
 seen MWI specifically mentioned as standard message sent via SS7, but
obviously
 it's being done.  I don't know enough about the details of SS7 to know
what
 messages can be sent or if there's a generic container message that can be
used
 for anything.
--- Some international implementations do this in more easy, but more
faulty... when the switchs make a call diversion for a VoiceMail system, its
starts to make a stutter tone, or send MWI indications to the phone. If the
subscriber dials to VoiceMailMain application to retrieve the messages, the
switch stops the stutter tone, or anything. I know, is a crap, but in some
coutries it was implemented in this way.

  From my research that's correct.  The PRI's D channel doesn't speak SS7,
 although the protocols are extremely similar in function.  Everything I've
read
 says that getting out-of-band signalling to the CP was the whole point in
 creating ISDN.  What I'm trying to find out is if there's some way to send
a D
 channel message that would get translated directly into SS7.  The ISUP
layer of
 the SS7 protocol is the ISDN User Part - is that designed to encapsulate
CLASS
 messages?
 No, you don't directly send information between PRI and ISUP
message... To understand correctly this, I send a complete ISUP trace.


 Take ANI for example.  Your PRI sends the ANI information to the near end
switch
 over the D channel which then passes it on (without verification, I might
add)
 on to the destination switch via SS7.  This is a case where the
information is
 transferred directly.  What about LIDB lookups or route information?  Is
there
 any way to get this, which is definitely available over SS7, from the D
channel?
 This is not 100% accured... In all implementations that I saw, you can
send any ANI to PSTN switch in PRI, but this switch makes a check to
determine if your ANI is valid, and if your ANI is your real ANI!!! This is
the most important check.

  So, the proper answer is that if you really want to implement this PRI -
SS7
  - PRI message, you should really be talking to your nearest CO Engineer
or
  Telco Enterprise Business Office where they handle this all the time for
  enterprise call center applications.
 Mhh... is really hard...

  On the other hand, maybe Gus could contribute a regular tutorial on how
he's
  got various things interconnected.  The more the info, the better.  Gus
once
  asked if we want the plethora of info he can provide.  I vote yes.
 Sure! I have some documents very useful. Please give me some ideas to
know what things you like.

Regards,

Gus


+-+-++
|BITMASK  |ID Name  |Comment or Value  
  |
+-+-++
|10/28 08:36:03,589  1:B (Rx):16  MSU  IAM  8971  16289  25  `5444f`  `6751010866` 
  |
|MTP Level 2 (MTP-L2)  MSU (= Message Signal Unit) 
  |
|Message Signal Unit   
  |
|-0101101 |Backward Sequence Number |45
  |
|1--- |Backward Indicator Bit   |1 
  |
|-0110100 |Forward Sequence Number  |52
  |
|0--- |Forward Indicator Bit|0 
  |
|--100110 |Length Indicator |38
  |
|00-- |Spare|0 
  |
|0101 |Service Indicator|ISDN User Part
  |
|--00 |Sub-Service: Priority|Spare/priority 0 (U.S.A. 
only)  |
|10-- |Sub-Service: Network Ind |National message  
  |
|**b14*** |Destination Point Code   |16289 
  |
|**b14*** |Originating Point Code   |8971  
  |
|CCITT BLUE BOOK ISDN User Part (ISUP)  IAM (= Initial Address)
  |
|Initial Address   
  |
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Re: [Asterisk-Users] dialogic support

[CW_ASN - Gus]

Yes, its true. Contact to [EMAIL PROTECTED]



- Original Message - 
From: tad [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 4:41 PM
Subject: [Asterisk-Users] dialogic support


 i am new to asterisk, and looking to develop an application using a
 dialogic card. as far as i can tell, drivers for these cards are
 available, but are not free. is that still true? if so, whom does one
 contact about licensing?
 
 thanks,
 tad
 
 
 
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Re: [Asterisk-Users] passing digits for voicemail from sip gateway

[CW_ASN - Gus]

What kind of gateway are you using? Did you set dtmf-relay in that gateway?

Regards,

Gus

- Original Message - 
From: Steve Dolloff [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 4:50 PM
Subject: [Asterisk-Users] passing digits for voicemail from sip gateway


I am seeing strange behavior that I don't understand.  Voicemail2 and
voicemailmain2 work fine if I call from a sip phone directly connected
to *, but if I call either of them from an analog line on the other side
of a sip gateway, voicemail seems to ignore digits.  If I am recording a
message and press #, nothing happens except that it records the tone
onto the message and I can't specify a mailbox using digits either, it
just hangs up on me.  Is this a config problem on the gateway?

Thanks,

Stephen


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Re: [Asterisk-Users] Groups in *

[CW_ASN - Gus]

Lars:

Anything you want is possible to do with Asterisk... the matter is how much
time you want to spend to build that applications... I think that is posible
to do that with AGI scripts...

Regards,

Gus

- Original Message -
From: Lars Fredriksson [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 4:52 PM
Subject: [Asterisk-Users] Groups in *


 Hi list!

 I have a little question about groups and Asterisk ... is there anyone out
there that can say if Asterisk can do any of this;

 We have a customer that want call handling we cant give him with a
traditional PBX, and I'm running Asterisk @home so I thought I could give it
a try ...

 The customer wants that incoming call should go to one group with some
phones in it, if the group is busy tha call should stay there for xxx
seconds before it goes to another group. But if there are phones free in the
group they should ring for xxx seconds before the call goes to another
group. And like this it would go on with lots of groups ;-)

 He also wants queue messages in all groups and the possibility for the
phones to log in and out of the different groups (in the morning one phone
should be member of three groups, and after lunch log out of those groups
and log on to another group ...)
 I think some kind of web-frontend would be quite kewl, so each employee
could log on to a webpage and mark which groups he will answer on (I don't
know how * keeps track of such things?)

 We have tried with PBX's like Panasonic TDA, Ericsson BusinessPhone, Avaya
INDeX, Avaya IPOffice and Siemens and none of those can do this ...

 Thanks for any answer!

 Best regards Lars Fredriksson, Sweden

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Re: [Asterisk-Users] Music Onhold Configuration

[CW_ASN - Gus]

MPG123 is not included in Asterisk...
Download the package:

http://www.mpg123.de/cgi-bin/sitexplorer.cgi?/mpg123/precompiled/

Install using:

rpm -ivh mpg123-0.59q-1.i386.rpm

Copy the file mpg123 from /usr/local/bin to /usr/bin

That's all...

Please read the posts, this issue was treated before.

Regards,

Gus

- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, October 27, 2003 4:52 PM
Subject: Re: [Asterisk-Users] Music Onhold Configuration



 You are right. I do not have mpg123 installed. Is it not included in
 Asterisk build? I would appreciate it if you could give some instructions
 on how to install this process.

 Thank you in advance,
 Kang




   Ing. Angel Gomez Garcia
   [EMAIL PROTECTED]To:
[EMAIL PROTECTED]
   Sent by:  cc:
   [EMAIL PROTECTED]Subject:  Re:
[Asterisk-Users] Music Onhold Configuration
   .digium.com


   10/22/2003 03:03 AM
   Please respond to
   asterisk-users







 Hi.

 Do you have mpg123 installed and mpg123 in /usr/bin ? I have moh
 working with snom200 and there was no issue to have them working.
 I even put an extension in my extensions.conf so the user can dial
 it an hear the music, cause the snom200 has call waiting they don't miss
 calls because of them hearing moh

 exten = 0400,1,Answer()
 exten = 0400,2,MusicOnHold(random)

 and musiconhold.conf has
 [classes]
 random = quietmp3:/var/lib/asterisk/mohmp3,-z

 Good luck.

 [EMAIL PROTECTED] wrote:

 Jean-Christophe,
 
 Thank you very much for your help. I configured the Music On Hold by
 following your sample, it seemed work fine by looking at the Trace. But
no
 Music came up on my SIP phone SNOM200. I checked /var/lib/asterisk/mohmp3
 and found only one MP3 file there sample-hold.mp3. Do you know what's
 wrong with it?
 
 Thank you in advance,
 Kang
 
 
 
 
 

   Jean-Christophe Heger

   [EMAIL PROTECTED]  To:
 [EMAIL PROTECTED]

   Sent by:  cc:

   [EMAIL PROTECTED]Subject:  Re:
 [Asterisk-Users] Music Onhold Configuration
   .digium.com

 

 

   10/20/2003 06:10 PM

   Please respond to

   asterisk-users

 

 

 
 
 
 
 /etc/asterisk/musiconhold.conf
 [classes]
 default = mp3:/var/lib/asterisk/mohmp3
 
 /etc/asterisk/extensions.conf
 exten = 101,1,Answer
 exten = 101,2,MusicOnHold(default)
 
 That's about what is said in the manual (RTFM ;-) and it works great.
 
 Jean-Christophe
 
 - Original Message -
 From: [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Cc: [EMAIL PROTECTED]
 Sent: Monday, October 20, 2003 10:56 PM
 Subject: [Asterisk-Users] Music Onhold Configuration
 
 
 
 
 Anyone can share me with Music Onhold Configuration sample?
 
 Thanks in advance for your help,
 Kang
 
 
 
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Re: [Asterisk-Users] Asterisk on FreeBSD

[CW_ASN]

Sometimes, if * dies, mpg123 keeps running and eats all memory.
Try to stop *, kill all mpg123 instances and try again.

Also, you can modify your start script to kill all mpg123 instances before *
starts 'killall -9 mpg123'

Regards,

Gus


- Original Message -
From: TeleSIP [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, October 26, 2003 6:41 PM
Subject: Re: [Asterisk-Users] Asterisk on FreeBSD



 - Original Message -
 From: Rich Adamson [EMAIL PROTECTED]
 To: [EMAIL PROTECTED]
 Sent: Sunday, October 26, 2003 3:11 PM
 Subject: Re: [Asterisk-Users] Asterisk on FreeBSD


   My Asterisk (fresh CVS) takes 98% of the system load on my FreeBSD
 server.
   On a slower CPU linux system, Asterisk runs at 0.1% - both without any
   active channels...
  
   Any ideas, anyone recognizing the problem?
 
  Is 'top' suggesting that * is actually consuming 98%?
 
  If it is, take a look at the * logs for signs of what it might be. We've
  seen this happen on a lab RH9 system, but its usually while we been
doing
  other unusual things. (In our case, two extra instances of mpg consuming
  the ~98%; copying *.conf files to a second system that didn't actually
  have any x100p cards in it, etc.)
 Same here with mpg123.  Once time we saw 2 extra mpg123 processes eating
99%
 of the CPU.  No idea why they were there.

 
  FWIW, I'm running yesterday's cvs on two RH9 systems just fine.
 
 
 
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